Huawei AR G3 Series Enterprise Routers V200R002C01...

Huawei AR G3 Series Enterprise Routers

V200R002C01

Voice Feature White Paper

Issue 01

Date 2012-06-10

HUAWEI TECHNOLOGIES CO., LTD.

Issue 01 (2012-06-10) Huawei Proprietary and Confidential

Copyright © Huawei Technologies Co., Ltd.

i

Copyright © Huawei Technologies Co., Ltd. 2012. All rights reserved.

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mailto:[email protected]


Voice Feature White Paper



ii

AR Voice Feature White Paper

Keywords

IP PBX, VoIP, SIP

Abstract

This document describes voice features supported by the AR G3 series enterprise routers.

Acronyms

Acronym Full Name

AR Access Router

IMS IP Multimedia Subsystem

VoIP Voice over Internet Protocol

SIP Session Initiation Protocol

IP PBX IP Private Branch eXchange

AG access gateway

FXO Foreign Exchange Office

FXS Foreign Exchange Station

SIPUE Sip user agent

POTS Plain Old Telephone Service

CDR Call Detail Record


Voice Feature White Paper Contents



iii

Contents

1 SIP AG Overview .......................................................................................................................... 1

2 IP PBX Overview ........................................................................................................................... 3

3 SIP .................................................................................................................................................... 6

3.1 SIP Structure..................................................................................................................................................... 8

3.2 SIP Messages .................................................................................................................................................. 10

3.3 User Registration Process ............................................................................................................................... 10

3.4 VoIP (SIP) MO Process .................................................................................................................................. 12

3.5 VoIP (SIP) MT Process ................................................................................................................................... 13

3.6 Call Release Process....................................................................................................................................... 14

3.7 FoIP (FAX over IP) ........................................................................................................................................ 15

3.7.1 FoIP Overview ...................................................................................................................................... 15

3.7.2 FoIP Transmission Mode ...................................................................................................................... 15

3.7.3 Low-Speed Fax and High-Speed Fax ................................................................................................... 17

3.8 MoIP (Modem over Internet Protocol) ........................................................................................................... 17

3.8.1 MoIP Connection Type ......................................................................................................................... 18

4 Basic IP PBX Services ................................................................................................................. 19

4.1 FXS Access .................................................................................................................................................... 19

4.2 FXO Access .................................................................................................................................................... 21

4.2.1 FXO as the Calling Party ...................................................................................................................... 21

4.2.2 FXO as the Called Party ........................................................................................................................ 22

4.3 E1/PRI Access ................................................................................................................................................ 23

4.3.1 ISDN Signaling ..................................................................................................................................... 23

4.3.2 Q.931 Call Instances ............................................................................................................................. 25

4.3.3 IP PBX Access to the PSTN .................................................................................................................. 27

4.4 SIP UE Access ................................................................................................................................................ 28

4.4.1 SIP UE Registration .............................................................................................................................. 29

4.4.2 Process of a Call Between SIP UEs ...................................................................................................... 30

4.5 Access to the IMS Through SIP ..................................................................................................................... 31

4.5.1 Access to the IMS Through SIP with Registration ................................................................................ 32

4.5.2 Calling and Called Parties Access to the IMS Through SIP.................................................................. 33

4.6 PBX Communication Through SIP ................................................................................................................ 34

4.7 Fax/Modem .................................................................................................................................................... 35


Voice Feature White Paper Contents



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4.8 Number Change ............................................................................................................................................. 35

4.9 Intelligent Routing ......................................................................................................................................... 36

4.10 CDR ............................................................................................................................................................. 38

5 IVR Service ................................................................................................................................... 39

5.1 Dialing an Extension Number ........................................................................................................................ 39

5.2 Triggering a Simultaneous Ringing Service ................................................................................................... 41

5.3 Triggering a Sequential Ringing Service ........................................................................................................ 43

5.4 Triggering the Line Selection Service ............................................................................................................ 45

5.5 Triggering Call Queuing................................................................................................................................. 47

6 BEST Function Description ....................................................................................................... 50

6.1 Overview ........................................................................................................................................................ 50

6.2 Description ..................................................................................................................................................... 50

7 Power Outage Survival .............................................................................................................. 52

8 Call Manager System.................................................................................................................. 53

8.1 Advantages ..................................................................................................................................................... 53

8.2 Deployment .................................................................................................................................................... 54

9 Other Services Supported by AR G3 Series Routers ........................................................... 55

10 SIP NAT Traversal .................................................................................................................... 59

10.1 Overview ...................................................................................................................................................... 59

10.2 SIP NAT Traversal Principles ....................................................................................................................... 60

10.3 AR SIP NAT Traversal Solution (SBC Solution) ......................................................................................... 61

11 AR Voice Solution .................................................................................................................... 63

11.1 AR Inter-Branch Voice Communication Solution ........................................................................................ 65

11.1.1 Centralized Call Control Model .......................................................................................................... 65

11.1.2 Distributed Call Control Model ........................................................................................................... 68

11.1.3 Hybrid Call Control Model ................................................................................................................. 71

11.2 AR Connecting to an IMS/NGN Network as AG ......................................................................................... 71

11.2.1 Market Positioning and Intended Customers ...................................................................................... 71

11.2.2 Network Topology and Solution ......................................................................................................... 72


Voice Feature White Paper 1 SIP AG Overview



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1 SIP AG Overview

Definition

SIP AG is a voice access gateway (AG) device based on the Session Initiation Protocol (SIP).

It is configured between the public switched telephone network (PSTN) and IP multimedia

subsystem (IMS), and is mainly used to convert signals between analog and digital forms.

Purpose

The emergence of the packet-switched network leads to revolutionary changes to the

telephony system. Many new technologies are also developed for this new bearer network.

The Voice over IP (VoIP) service enables IP networks to carry voice services (such as

traditional telephone services). In addition, the new IMS provides powerful support for VoIP

application. An IMS network is a standard next-generation carrier network that provides

mobile or fixed-line multimedia services. It supports traditional packet switched and circuit

switched telephony systems. Compared with the traditional PSTN, the IP bearer network

features higher resource utilization and shared lines for calls. Currently, the VoIP technology

has been put into commercial use.

Traditional circuit switched telephone networks have been developing for years and a large

number of devices are still in service now. Replacement of existing telephone networks with

IP bearer networks can cost too much. SIP AGs can be used to connect the voice network and

data network cost-effectively. Huawei AR G3 series routers can function as SIP AGs to

connect the PSTN network and IP data network.

As shown in Figure 1-1, ARs serve as SIP AGs to integrate the voice network and the IP data

network based on the SIP protocol.


Voice Feature White Paper 1 SIP AG Overview



2

Figure 1-1 Typical networking where the SIP AG functions as the voice gateway

SIPAG

POTS Modem FAX

IP

Network

POTS

IMS

SIPSIP

SIPAG

Benefits

Although the VoIP service shares bandwidth with other services on the Ethernet, proper

network planning and quality of service (QoS) configuration ensure high quality of enterprise

voice services.

The use of AR G3 series routers as the SIP AGs to provide VoIP services brings the following

benefits to enterprises:

Low costs: Traditional calls and fax services use circuit switched mode and occupy

communications lines exclusively. Long distance call and fax services are expensive. The

VoIP service with SIP AGs serving as voice gateways can reduce the communication

costs for enterprises.

High call quality: SIP AGs ensure call completion rate, voice quality, and service types

by configuring QoS.

Smooth upgrade/capacity expansion: A VoIP system is compatible with the existing

telephony systems and office platforms, and the service capacity can be increased when

the enterprise scale expands.


Voice Feature White Paper 2 IP PBX Overview



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2 IP PBX Overview

Definition

A private branch exchange (PBX) is a telephone exchange that serves a particular business or

office. An IP-based PBX (IP PBX) is the server used on the internal IP telephone network of

an enterprise for call control and configuration management.

Purpose

VoIP technology converts analog voice signals to digital signals, encapsulates digital signals

in IP data packets, and transmits IP data packets on the IP data network in real time. By using

the Internet, VoIP provides more and better services than the traditional PBX. For example,

VoIP can transmit voice, fax, video, and data services on the IP network with low costs. VoIP

provides unified messaging, virtual phone, virtual voice/fax email, number query, Internet call

center, Internet call management, video conference, ecommerce, fax S/F, and store and

forward of other information.

Traditional PBXs exchange calls inside an enterprise and between the enterprise network and

the PSTN. One PBX integrates the telephone, fax, and modem functions. PBXs are widely

used in enterprise offices and greatly enhance enterprise efficiency. However, traditional

PBXs cannot meet the requirements for computer telephony integration (CTI) and VoIP. In

addition, these PBXs are expensive and do not use standard and open platforms, making the

interconnection between PBXs of different vendors difficult. IP PBXs provide local exchange

and IP user access functions. AR G3 series routers can function as IP PBXs to integrate voice

communications into enterprise data networks so that an integrated voice and data network is

established to connect offices and employees around the world. AR G3 series routers can also

connect to traditional POTS phones through voice gateways, making voice networks scalable.

Benefits

Compared with traditional PBXs, AR G3 series routers provide the following benefits to

enterprises when functioning as IP PBXs:

Low construction costs: IP PBXs can be deployed on the existing IP network of an

enterprise, saving the costs on constructing and maintaining multiple networks.

Low management costs: IP PBXs simplify the process to add, replace, or remove a

terminal. For example, an IP phone can be moved by simply connecting the phone to

another network interface. Unlike a traditional PBX, an IP PBX does not require

additional configuration for the moved IP phone.





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High work efficiency: IP PBXs can rapidly integrate multiple related systems so that

enterprises do not need to deploy single-function systems.

Highly reliable communication: IP PBXs ensure normal provisioning of internal

services when egress transmission channels of an enterprise fail.

Flexible solution: IP PBXs can be deployed in distributed networking to meet

requirements of IP-based voice and data communication. This distributed networking

allows enterprises to construct enterprise networks cross the cities, provinces, and even

countries.

Self-service maintenance: IP PBXs provide an individual service management system for

enterprises and helps reduce carriers' maintenance costs. For example, an IP PBX

provides extension number selection, short number self-planning, toll call right

modification, number portability, and internal line addition.

Customized development: To improve work and communication efficiency, IP PBXs can

integrate the enterprise OA process, enterprise address book, and the click-to-dial

function based on enterprises' needs.

Featured solution: IP PBXs provide featured application solutions such as hotel

telephone service and voice record.

Abundant ICT applications: IP PBXs can be integrated with the UC system to enrich the

ICT applications of enterprises.

High resource utilization efficiency: An IP PBX on a local area network (LAN) manages

the computer network and telephone network effectively based on actual conditions and

implements resource sharing.





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Figure 2-1 shows a typical IP PBX networking.

Figure 2-1 Typical IP PBX networking

SIPPOTSFAX

FAX

POTS

IAD

SIP AG

FAX

POTS

POTS

FAX

POTS

POTS

SIP

SIP

SIP

IAD

IAD

FAXPOTSPOTS

TDM PBX

IP PBX

E1 Ethernet

SIP AG

SIP

SIP

SIP

VOICE

SIPPOTSFAX

FAX

POTS

IAD

IP PBX

VOICE

VOICE

VOICE

HeadquartersNewly built area

New built&migrated area

Migrated area

Access

switch

Aggregation

switch

FXS (RJ11 telephone line)

Branch(Centralized

call control)

Branch

(Distributed call

control)


Voice Feature White Paper 3 SIP



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3 SIP

The Session Initiation Protocol (SIP) is an application-layer protocol used to create, modify,

and terminate multimedia sessions. Multimedia sessions are used for applications such as

multimedia conferences, remote education, and Internet calls. SIP can be used to initiate a

session and to invite members to the session established in other ways (for example,

multi-party conference). SIP transparently supports name mapping and redirection services to

implement ISDN, intelligent network (IN), and personal mobility services.

Once a session is set up, media streams are directly transmitted at the bearer layer using the

Real-Time Transport Protocol (RTP). SIP, proposed by the Internet Engineering Task Force

(IETF) in 1999, is a signaling protocol implementing real-time communication on an IP

network.

SIP supports the following functions for establishing and terminating multimedia

communications:

1. User location: determines the end system used for communication.

2. User availability: determines the media and media parameters to be used in

communication.

3. User availability: determines the willingness of the called party to engage in

communication.

4. Session setup: establishes session parameters for the called and calling parties.

5. Session management: includes transfer and termination of sessions.





7

SIP is designed as part of the overall IETF multimedia data and control architecture, as shown

in Figure 3-1.

Figure 3-1 IETF multimedia data and control architecture

H.323 SIP RTSP RSVP RTCP

H.263 etc.

RTP

TCP UDP

IP

PPP PPPAAL3/4 AAL5

Sonet ATM Ethernet V.34

SIP is used with other protocols. For example, the Resource Reservation Protocol (RSVP)

reserves network resources, the Real-Time Transport Protocol (RTP) transports real-time data

and provides QoS feedback, the Real-time Stream Protocol (RTSP) controls delivery of

streaming media, the Session Announcement Protocol (SAP) advertises multimedia sessions

in multicast mode, and the Session Description Protocol (SDP) described multimedia sessions.

However, the functionality and operation of SIP do not depend on any of these protocols.

SIP can also be used with other session setup and signaling protocols. In that mode, an end

system uses SIP to determine an appropriate end system address and protocol from a given

address that is protocol-independent. For example, SIP can be used to determine whether the

local end can communicate with the peer through H.323. If so, SIP obtains the H.245 gateway

address and user address, and then uses H.225.0 to establish the call. In another example, SIP

can be used to determine whether the called party is connected through the PSTN and specify

the called number. It is recommended that the Internet-to-PSTN gateway be used to establish

the call.

SIP fundamentally changes the communications service provisioning mode and the

consumption habits of communications users. Services such as video and audio calls,

messaging, web, email, synchronous browse, and conference services are integrated, bringing

innovations to the telecommunication industry. SIP has the following advantages as a control

layer protocol:

1. Based on open Internet standards, SIP is suitable for integration of voice and data

services, and can implement call control across media and devices. In addition, SIP

supports various media formats and can dynamically add or delete media streams, so that

various services can be deployed easily.

2. Extends the intelligent network to service side and end systems, reducing the network

burden and facilitating service development

3. Supports application-layer portability functions, including dynamic registration, location

management, and redirection management.





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4. Provides presence/Fork/Subscription features, which facilitate new service development

5. It is simple and scalable.

3.1 SIP Structure

The SIP protocol logically consists of the following elements:

User agent: also called the SIP terminal. It is the end user of the SIP system and is

defined as an application in RFC3261. Based on roles in a session, user agents can be

classified into the user agent client (UAC) and user agent server (UAS). The UAC

initiates a call request, and the UAS responds to the call request.

SIP proxy server: an intermediate device. It can function as a server to parse user names

and function as a client agent to initiate a call request to the next-hop server, which then

determines the next hop address.

SIP register server: an important part in the SIP system. It receives user registration

information and maintains the information into the address database.

Location server: stores and returns user address information. It obtains address

information from the register server or other databases, and then uploads the address

registration information to the location server.

Redirect server: determines paths of call. After obtaining the next hop address, this

server requests the previous-hop user to initiate a request directly to the next hop. At the

same time, this server stops controlling the call. For example, if Bob wants to call Lara

and this request is sent to the redirect server. The redirect server obtains the address of

Lara and returns the address to Bob. Then, Bob can resend the session invitation to the

address.

Actually, functions of the preceding SIP servers are provided by one server. They are only

identified logically. The following figures show interactions between the SIP components.

Interaction between the UA, register server, and location server: registration

Register Server

This is 010-8888.

I am at

[email protected].

010-8888 is at

[email protected].

Location ServerI have made a record.OK. The registration completes.

1 2

34UA





9

Interaction between the UA, proxy server, and location server: call routing

Proxy Server

I want to chat with UA2.

Location Server

1

UA1

Add

ress

of U

A2

2

Whe

re is

UA2?

3

UA1 is asking for you.

4

5Hello.

6Hi.

Is it Tom? This is Jerry.

7

UA2

Interaction between the UA, redirect server, and location server: call redirection I w

an

t to c

ha

t with

UA

2.

Location Server

1

UA1

Where is UA2?

2

Address of UA23

Th

is is

the

a

dd

ress o

f UA

2.

4

5

Hello. Is it Tom? This is Jerry.

6Hi, Jerry.

7

RTP

Redirect Server

UA2





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3.2 SIP Messages

SIP messages are encoded in text format. There are two types of SIP messages: request and

response.

RFC 3261 defines the following SIP request messages:

INVITE: invites a user to a call.

ACK: acknowledges a response message.

OPTIONS: negotiates communication capabilities with the peer.

BYE: terminates a session.

CANCEL: cancels a session establishment.

REGISTER: registers user location information with a registrar server.

SIP response messages are sent in response to request messages, informing calling parties of

call or registration results. Status codes identify the types of response messages. A status code

is a 3-digit integer. The leftmost digit indicates the response message type, and the other two

digits provide additional information, such as how a received request message is processed.

RFC 3261 defines the following status codes:

100 to 199: provisional. A request has been received and is being processed.

200 to 299: success. A request has been successfully processed.

300 to 399: redirection. Further action needs to be taken to complete the request.

400 to 499: client error. A request contains incorrect syntax or cannot be processed by the

server.

500 to 599: server error. The server failed to process a valid request.

600 to 699: global failure. A request cannot be processed by any servers.

3.3 User Registration Process

Before a SIP user initiates a call, the user must register user information (for example,

mapping between the domain name and the IP address) on the home network. The registration

process can be implemented in non-authentication mode or authentication mode. After the

system is powered on or a user is added, the user registration process starts.





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Registration Process in Non-authentication Mode

Figure 3-2 Registration process in non-authentication mode

SIP AG IMS Core

Register

Response 200

As shown in Figure 3-2, the SIP AG sends a Register message to the IMS Core for each user.

The Register message contains information such as the user identity. When receiving the

Register message, the IMS Core checks whether the user is configured in the IMS. If the user

is configured, the IMS Core returns a Response-200 message to the SIP AG. If the user is not

configured, the IMS Core returns a Response-403 message to reject the registration.

The AR supports individual registration and group registration. In individual registration

mode, users register on the IMS core individually through SIP AT0 trunks. In group

registration mode, multiple users can register on the IMS core together, which reduces the

number of register messages and avoids registration storms.

Registration Process in Authentication Mode

Figure 3-3 Registration process in authentication mode

SIP AG IMS Core

Register

Response 401/407

Register

Response 200





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As shown in Figure 3-3, the SIP AG sends a Register message to the IMS Core for each user.

The Register message contains information such as the user identity.

When receiving the Register message, the IMS Core queries and learns that this SIP AG

registration requires authentication. Then, the IMS Core returns Response-401/407, which

contains information such as the key and encryption method. The SIP AG encrypts the user

name and password with the key, and sends them in a Register message to the IMS Core. The

IMS Core decrypts the Register message and checks whether the user name and password are

correct. If they are correct, the IMS Core returns Response-200.

The AR supports the DIGEST MD5, DIGEST MD5-SESS, and AkAv1-MD5 algorithms for

authentication and encryption.

3.4 VoIP (SIP) MO Process

Figure 3-4 shows the VoIP (SIP) call process on the calling party side.

Figure 3-4 VoIP MO process

200(callee offhook)

Caller offhook

dialtone

P1

1st digit

P2

P4

P5

P6

P3

Dialtone stopped

2st digit

3st digit

D1:1NVITE(SDP)

D2:100 Trying

D3:180 Ringing

D4:200 OK

D5:ACKconversation

USER1 AG P-CSCF-O

IMS

Network





13

P1: The AG receives the pick-up message from the calling party and plays the dial tone

for the calling party.

P2: When receiving the first dial number, the AG stops the dial tone and matches the

number with the digitmaps.

P3: After receiving N numbers, the AG detects that the numbers match a digitmap. Then

the AG constructs an Invite message and sends it to the P-CSCF.

P4: When receiving 100 Trying, the AG learns that the peer has received the Invite

message. Then the AG stops the process of retransmitting the Invite message.

P5: The AG receives 180 Ringing, indicating that the phone of the called party rings. The

AG plays the RBT for the calling party.

P6: The AG receives 200 OK message, indicating that the called party has picked up the

phone. Then the AG stops playing the RBT and changes the flow mode to bidirectional.

The AG constructs an ACK message to the P-CSCF.

Besides normal calls, there are other scenarios. When the calling party initiates a call, the

P-CSCF performs either of the following operations:

If the data about the calling party exists but is not registered, the P-CSCF rejects the call

from the calling party and returns message 403.

If there is no data about the calling party, the P-CSCF rejects the call from the calling

party and returns message 404.

3.5 VoIP (SIP) MT Process

Figure 3-5 shows the VoIP (SIP) call process on the called party side.

Figure 3-5 VoIP (SIP) MT process

ring

P1

P2

P3

Callee offhook

D2:100 Trying

D5:ACK

conversation

D1:INVITE(SDP)

D3:180 Ringing

D4:200 OK

IMS

Network

USER1 AG P-CSCF-T





14

P1: After receiving an INVITE message from the P-CSCF, the AG constructs a 100

Trying message and sends it to the P-CSCF. The AG locates the called party according to

the P-Called-Party-ID header field, RequestURI, and TO header field carried in the

INVITE message. If the TEL-URI field is used, the header fields can be not used. The

AG can locate the called party according to the phone number in the TEL-URI field.

Then the AG plays the ring tone to the called party. The AG constructs a 180 Ringing

message and sends it to the P-CSCF, notifying that the phone of the called party is

ringing.

P2: After receiving the off-hook message from the called party, the AG stops ringing. In

addition, the AG constructs a 200 OK message and sends it to the P-CSCF, notifying the

called party has picked up the phone.

P3: The AG receives an ACK message and the calling party and the called party talk with

each other.

Besides normal calls, there are other scenarios. When receiving the Invite message, the AG

performs either of the following operations:

If the data about the called party exists but is not registered, the AG rejects the call from

the calling party and returns message 403 to P-CSCF.

If there is no data about the called party, the AG rejects the call from the calling party

and returns message 404 to P-CSCF.

3.6 Call Release Process

Figure 3-6 shows the VoIP (SIP) call release process.

Figure 3-6 Call release process

IMS

Network

USER1 AG P-CSCF-O

P1

P2

onhook

conversation

D2:200 OK

D1:BYE





15

P1: After receiving the onhook message of the user, the AG constructs a BYE message

and sends it to the P-CSCF to release the DSP resource allocated to the user.

P2: After receiving 200 OK from the P-CSCF, the AG releases the call.

3.7 FoIP (FAX over IP)

Traditional fax is sent and received through the PSTN. Fax services are widely used because

various types of information can be easily transmitted at a high speed.

The International Telegraph and Telephone Consultative Committee (CCITT) defines four fax

machine standards, namely, G1, G2, G3, and G4 fax machines.

G1: low-speed analog fax machines using analog frequency shift keying signals, and in

black and white

G2: medium-speed analog fax machines using analog phase shift keying signals in black

and white; compressed frequency band at a transmission speed double that of G1

G3: high-speed digital fax machines using modulating signals in black and white at a

transmission speed four times that of G1

G4: high-speed digital fax machines for the ISDN network at a speed of 64 kbit/s, using

hybrid fax and telegraph terminals

Due to the limitation of speeds or cables, G1, G2, and G4 fax machines are not widely used.

Only G3 fax machines are commonly used for fax communication. G3 fax machines use a

digital signal processing technology. Image signals are digitalized and compressed in a fax

machine, converted to analog signals by a modem, and finally transmitted to a PSTN switch

through common subscriber lines.

3.7.1 FoIP Overview

Fax over IP (FoIP) sends and receives fax over the Internet. Compared with traditional fax,

FoIP has the following benefits:

Low fee: FoIP fully use the worldwide deployment and low communication fees of the

Internet, and significantly reduces fax fees for enterprises.

High security and QoS: FoIP uses advanced transmission and encryption technologies to

improve the content definition and confidentiality, which are better than those of the

traditional fax and IP telephone fax.

High intelligence: FoIP automatically resends fax in a specified period and returns

success or failure information to the user's email box.

3.7.2 FoIP Transmission Mode

FoIP supports two transmission modes (pass-through and T.38) and two switching modes

(auto-switch and initiated negotiation switch). That is, four fax modes are available:

auto-switch pass-through, auto-switch T.38, negotiation pass-through, and negotiation T.38.

Auto-switch: The AG detects fax signals and selects the transparent or T.38 mode based on

the configuration. In this case, the AG does not need to send any signal to the peer end.

Initiated negotiation: The AG detects fax signals, and then sends a REINVITE message

carrying negotiation parameters to negotiate the codec mode with the peer based on the

configuration.





16

In pass-through fax mode, fax data from the PSTN is modulated and then forwarded over an

end-to-end (E2E) voice tunnel on the IP network. The AG functions as the gateway between

the PSTN and the IP network and does not participate in modulation or demodulation. The

AG is used as the gateway and fax machine to forward voice flows. Fax can be transmitted

using pre-configured voice codes. Alternatively, the gateway automatically switches to the

high speed coding mode of G.711. Compression loss of fax signals is relatively large when the

G.729 protocol is used, and fax signals may not be demodulated correctly at the peer end.

Therefore, the G.711 protocol, which causes less compression loss, is usually used for fax

pass-through. Figure 3-7 shows the data forwarding process of fax pass-through.

Figure 3-7 Data forwarding process of pass-through

FAX FAX

Gateway

T.30 signaling

Gateway

Fax analog data Fax analog data

Analog data passes a VoIP

tunnel at a rate of 64 kbit/s.

IP network

G.711 coding

64 kbit/s

G.711 coding

64 kbit/s

During a T.38 fax call, the sending gateway demodulates a T.30 fax sent from the PSTN. The

demodulated fax data is encapsulated in datagrams and sent to the receiver across the IP

network. The receiver gateway modulates the datagrams into T.30 fax data and sends the fax

data to the receiver. Figure 3-8 shows the data forwarding process.

Figure 3-8 Data forwarding process of T.38 fax relay

FAX FAX

Gateway Gateway

Fax analog data Fax analog data

Data packet transmission

IP network

DSP demodulation DSP demodulation

T.38 signalingT.30 signaling T.30 signaling





17

3.7.3 Low-Speed Fax and High-Speed Fax

Differences between low-speed fax and high-speed fax are as follows:

Standard: High-speed fax uses the V.8 data transmission process. Low-speed fax uses the

fax process defined by T.30. In addition, some low-speed fax terminals may use earlier

standards.

Rate range: Rate supported by high-speed fax ranges from 2.4 kbit/s to 33.6 kbit/s and

that supported by low-speed fax ranges from 2.4 kbit/s to 14.4 kbit/s.

Uplink transmission mode: High-speed fax uses only the pass-through mode. That is, fax

is transmitted at a high rate from a modem to a gateway. Low-speed fax uses

pass-through or T.38 mode. (T.38 mode does not support the rate of high-speed fax.)

Error correction mode (ECM) requirement: High speed-fax must use the ECM

mode, which is optional for low-speed fax.

DSP EC requirement: High-speed fax requires DSP EC to be disabled (because it has an

echo processing mechanism). Low-speed fax requires EC to be enabled (because it has

no echo processing mechanism).

3.8 MoIP (Modem over Internet Protocol)

A modulator demodulator (modem) is a device that is installed between a personal computer

(PC) and a telephone to convert signals exchanged between them. A PC transmits digital

signals to the modem port. The modem receives the signals and coverts (modulates) them into

analog signals. Then, the signals are processed as normal voice signals in the telephony

system. Signals sent from a telephone to a PC are processed reversely: Analog signals are

transmitted over telephone lines to a modem, which converts the analog signals to digital

signals, and sends the digital signals to a PC through the modem port.

Modems are used for signal format conversion, including analog to digital conversion and

digital to analog conversion. Other functions of a modem are as follows:

Coverts signal frequency domain, such as the conversion from low-frequency signals to

high-frequency signals and the modulation from digital baseband transmission to analog

channel transmission.

Extracts low-frequency signals from high-frequency signals.

Demodulates digital baseband signals.

Compresses network transmission data.

Controls coding and error correction.

MoIP provides modem services on the IP network or between the IP network and traditional

PSTN network.





18

3.8.1 MoIP Connection Type

Same as VoIP, MoIP supports the gateway-based PSTN-IP-PSTN network structure and

PSTN-IP network structure.

Modem communication includes modulation at the physical layer, error correction at the link

layer, and data compression at upper layers. Based on the ways gateways process data of

different layers, the following MoIP connection types are available:

0: Gateways do not process signals. Modulated signals are transparently transmitted on

the IP network through VoIP channels.

1: Gateways modulate modem signals but do not perform error correction or

compression, which are performed end to end by terminals.

2: Gateways modulate modem signals and correct errors, but do not compress data.

3: Gateways on both sides modulate signals, correct errors, and compress data. That is, a

gateway decompresses and recompresses modem signals, and then sends signals to the IP

network. The other gateway performs conversely.

4: Gateways modulate modem signals, correct errors, and compress data. Each gateway

is responsible for the compression and error correction at a certain direction.

The SIP-based modem can also adopt the auto-switch and initiated negotiation modes.

Transparent transmission modems working in initiated negotiation mode can be indicated by

one of the following types:

a=Modem: This transparent transmission modem mode using G.711 is proposed by

China Telecom.

a=silenceSupp:off: This transparent transmission modem mode using G.711 is proposed

in draft-ietf-sipping-realtimefax-01.txt.

a=gpmd:99 vbd=yes: The support of voice band data (VBD) is defined in V.152.


Voice Feature White Paper 4 Basic IP PBX Services



19

4 Basic IP PBX Services

IP PBX uses an integrated communication system. Through the telecommunications network

and Internet, only voice, fax, data, and video services can be provided by a single device. A

middle- or small-scale call center can be established, with low costs. By using the network

software and hardware, IP PBX improves the working efficiency and saves communication

costs.

4.1 FXS Access

Foreign exchange station (FXS) access is analog access. FXS implements connection between

a PSTN network and an IP network under the IP PBX architecture and provides PSTN

services.

Figure 4-1 shows the FXS implementation.





20

Figure 4-1 Implementation of FXS user access

User A (POTS) PBX User B (POTS)

Pick up the phone

Play the dail tone

After the initial ring tone is sent, the calling number is

displayed, and the ring tone is sent.

The calling

number is

displayed after

the ring tone.

Send the initial ring tone

Send the calling number

Play the ring tone

Enable the hangup transmission

Send the calling number

Play the ring tone

Play the ringback tone

Dail number (digits)

Pick up the phone

Stop playing the ring toneStop playing the

ringback tone

Call is set up.

Hang up the phone

Play the busy tone

Hang up the phone

Call is ended.

1. User A picks up the phone and the IP PBX plays the dial tone for user A.

2. User A dials the number of user B. After receiving the first digit, the IP PBX stops

playing the dial tone and starts analyzing the number.

3. After locating user B (the called party), the IP PBX sends the ring tone to user B. If the

IP PBX needs to send the calling number, the IP PBX will send the initial ring tone first

and then send the calling number. Therefore, the number of user A is displayed on user

B's phone.

4. User A hears the ringback tone.

5. User B picks up the phone. The IP PBX stops playing the RBT to user A, and stops

playing the RBT to user B. Then the call is set up between user A and user B.

6. After user A or user B hangs up the phone, the IP PBX plays the busy tone to the other

party. The call is ended.





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4.2 FXO Access

A foreign exchange office (FXO) accesses a PSTN network through a narrowband port and

common twisted pairs.

4.2.1 FXO as the Calling Party

Figure 4-2 shows the implementation of a call in which the user on the FXO port functions as

the calling party.

Figure 4-2 Implementation of a call in which the user on the FXO port functions as the calling

party

(POTS) User A PBX (AT0) AN

OffHook

Play DialTone

Dial Num

Number analysis

succeeds

OffHook

Play DialTone

Dial Num

Play Ringback Tone

Play Ringback Tone

Session is

set up

Called user picks

up the phone

1. The calling POTS user picks up the phone, hears the dial tone, and dials the called

number.

2. The IP PBX analyzes the number and finds that the outgoing call is made through the

FXO port. Then the IP PBX simulates offhook.

3. The IP PBX plays a dial tone to the FXO port, and determines whether to add a call

prefix to the called number according to the configuration.





22

4. If a prefix needs to be added, the IP PBX sends the configured prefix to the switch, and

several seconds later sends the called number to the AN and performs step 6.

5. If no call prefix needs to be added, the PBX sends the called number to the AN.

6. The AN analyzes the received number to locate the called POTS user, and plays the

ringback tone to the FXO port.

7. If the called POTS user picks up the phone, the AN sends a polarity reversal signal to the

FXO port. The calling and called POTS users start the conversation.

8. The calling or called party hangs up to end the call.

4.2.2 FXO as the Called Party

Figure 4-3 shows the implementation of a call in which the user on the FXO port functions as

the called party.

Figure 4-3 Implementation of a call in which the user on the FXO port functions as the called

party

(POTS) User A PBX (AT0) AN

Init Ring

Send caller number

Number

analysis

Local user Ring

Play Ringback Tone

IVR service Play SecDial Tone

Second dialing

Second number analysis

Ring

Play Ringback Tone

OffHook

Stop Ringback Tone

OffHook

1. The FXO port detects a ring message and sends the message to the IP PBX. The PBX

sends an off-hook message to the AN.

2. If the number bound to the FXO port is the number of a local user on the IP PBX, the IP

PBX analyzes the number to locate the called party and performs step 5.





23

3. If the number bound to the FXO port is an IVR number of the IP PBX, the IP PBX plays

the two-stage dial tone to the calling party through the FXO port and waits for the calling

party to dial the extension number.

4. The calling party dials the extension number. The PBX analyzes the called number to

locate the called party and performs step 6.

5. The IP PBX sends a ring message to the called party and plays the RBT to the calling

party through the FXO port.

6. After the called party picks up the phone, the PBX stops the ringback tone. The calling

and called parties start the conversation.

7. The calling or called party hangs up to end the call.

4.3 E1/PRI Access

4.3.1 ISDN Signaling

The Integrated Services Digital Network (ISDN) is a set of international communications

standards for digital telephone networks, and a typical circuit-switched telephone network

system.

ISDN supports various services including calls, video phones, data communication, and video

conferences by transmitting and processing voices, faxes, data, and images on a unified digital

network. Before the emergence of broadband access, ISDN is widely used for high speed

network access because of its faster speed than dial-up access. A relatively comprehensive

ISDN network is deployed in many areas.

ISDN can be classified into narrowband ISDN and broadband ISDN. Narrowband ISDN uses

the basic rate interface (BRI, 2B+D, 144 kbit/s) and primary rate interface (PRI, 30B+D, 2

Mbit/s). The BRI includes two 64 kbit/s bearer channels (B channels) and one 16 kbit/s

signaling channel (D channel or delta channel). The B channels are used for the transmission

of voice, data, and image, and the D channel is used for the transmission of signaling and

packet information.

With the emergence and wide application of VoIP, VoIP gateways are required to process

ISDN signaling messages. These messages and their functions are defined in ITU-T

Recommendation Q.931. Q.931, the network layer protocol of the telecommunication system,

mainly sets up and maintains calls on the ISDN, and terminates the logical network

connection between two devices.

ISDN Q.931 messages manage the connection on ISDN B channels. These messages can also

be modified and used in the Frame Relay and ATM UNIs to set up calls, or be used on NNIs

to provide services between networks. These messages are listed in Table 4-1 and brief

introductions to certain major messages are provided.

Table 4-1 ISDN layer 3 messages

ISDN Layer 3 Messages

NOTE

Message application varies with vendors and countries.

Call establishment messages HOLD

NOTIFY HOLD ACKNOWLEDGE





24

CALL PROCEEDING HOLD REJECT

CONNECT USER INFORMATION

CONNECT ACKNOWLEDGE Miscellaneous messages

PROGRESS CANCEL

SETUP CANCEL ACKNOWLEDGE

SETUP ACKNOWLEDGE CANCEL REJECT

DETACH CONGESTION CONTROL

DISCONNECT FACILITY

RELEASE FACILITY ACKNOWLEDGE

RELEASE COMPLETE FACILITY REJECT

RESTART INFORMATION

RESTART ACK REGISTER

Call information phase messages REGISTER ACKNOWLEDGE

RETRIEVE REGISTER REJECT

RETRIEVE ACKNOWLEDGE STATUS

RETRIEVE REJECT STATUS INQUIRY

NOTIFY: indicates that the called party is notified and the call is in process. This is the

response message to a SET UP message. After the called exchange sends ALERTING to

the called party, this message is sent from the called party to the calling party.

CALL PROCEEDING: sent to the call initiator, indicating that the call establishment has

started. This message also indicates that all mandatory messages for call establishment

are received and no more call establishment messages are accepted. During ISDN

implementation, this message is sent only on the setup initiator side.

CONGESTION CONTROL: used only in a USER INFORMATION message. This

message is used to manage the USER INFORMATION message stream. This message is

seldom used.

CONNECT: invoked faster when the called party picks up the phone. This message is

sent from the called party to the calling party, indicating that the call is received by the

called party.

CONNECT ACK: a response message to CONNECT. This message indicates that the

calling party and called party are authorized to participate in a call.

DISCONNECT: sent when the calling party or called party hangs up the phone. When

this message is sent, the E2E connection on the network is disconnected. Resources

reserved for the connection can be used by other calls.

INFORMATION: sent for more connection-related information by the user or network.

For example, an exchange can invoke this message to provide another exchange with

additional information about the connection.

BULLETIN: used only when a user or network provides connection-related information.





25

PROGRESS: part of the call setup process and not used in typical implementations. This

message can be used to indicate a call progress. It is used when interaction is required or

the exchange must provide inband information.

RELEASE: used when a DISCONNECT message is received. This message is sent by

the network or user to the receiver, indicating that the circuit reserved for the connection

is disconnected on the device.

RELEASE COMPLETE: a response message to RELEASE. This message indicates that

the sender has released the circuit, call reference number, and connection-related

resources. The RELEASE and RELEASE COMPLETE messages indicate that the

circuit is disconnected, resources are available for other calls, and the call reference

number is invalid.

RETRIEVE: used in relatively simple operations when a held call needs to be restored

on the network. The operation configuration varies with network providers. The basic

principle is that a user can change the idea when holding a call in a short period.

RETRIEVE ACK: a response message to RETRIEVE. This message indicates that the

request for restoring a held call is complete.

RETRIEVE REJECT: sent by the network, indicating that the request for restoring a held

call fails.

SETUP: used to start a call establishment. This message includes more information units

than any other Q.391 messages. The calling party always sends this message to the

network. In addition, the network always sends this message to the called party.

SETUP ACK: a response message to SETUP, indicating that the SETUP message is

received correctly. This message indicates that the call establishment process has started,

or more information is required to complete the call. In the later case, the receiver of the

SETUP ACK message must send additional information in an INFORMATION message.

STATUS: a response message to STATUS INQUIRY. This message may also be

sent when certain error occurs on a network node.

STATUS INQUIRY: sent by a user or network to query the status of a proceeding

operation, such as, a proceeding call. The STATUS and STATUS INQUIRY messages

can be flexibly implemented.

The ISDN allows call hold. Hold causes are not defined in specifications. The Q.931 protocol

provides the following messages for the operation:

HOLD: sent by a user to request the network to hold a call. This message can only be

initiated by a user instead of the network due to the transmission direction limit.

HOLD ACK: a response message to HOLD, indicating that a call is held.

HOLD REJECT: also a response message to HOLD, indicating that a call cannot be held.

USER INFORMATION: different from the INFORMATION message. This message

contains the User-User field that is not contained in an INFORMATION message.

FACILITY: sent by a user or network, providing supplementary call-related information,

such as keyboard information and displaying information.

RESTART: sent by a user or network to request connection restart. When this message is

sent, the identification channel returns to the idle status.

RESTART ACK: a response message to Restart.

4.3.2 Q.931 Call Instances

Figure 4-4 shows a call that is set up with Q.931 messages. Both users use traditional

telephones connecting to the ISDN terminal (the calling terminal and called terminal in Figure

4-4). The exchange terminal (ET) locates in the central office. The calling party picks up the





26

phone and dials the number of the called party. When receiving the Off-hook & Dial message,

the calling terminal sends a SETUP message to the local ET through an ISDN line. The ET

sends a SETUP ACK and starts to connect to the next ET. Interactions between two ETs are

indicated in dotted lines. The SETUP ACK and INFORMATION messages are optional. The

local ET sends a CALL PROCEEDING message to the calling terminal, indicating that the

call is in process.

Figure 4-4 Examples of ISDN signaling

User ACalling terminal User B

Called terminalET ET

CONNECT

SETUP

SETUP

Ringing

SETUP ACK

INFO CALL

PROCEEDING

ALERTING

ALERTING

Ring back

indication

Stop

Ring backCONNECT

ACK

CONNECT

CONNECTACK

Ongoing

connection

Hang-up

DISCONNECT

DISCONNECT

Off-hook

Pick-up

RELEASE

RELEASECOMPLETE

RELEASE

RELEASECOMPLETE

Off-hook&Dial

After receiving the SETUP message from the peer ET, the called terminal checks the message

and determines the called party and service type. Then, the called terminal checks the called

party line. If the line is idle, the terminal sends an ALERTING message to the called party.

When sending an ALERTING message, the called terminal also sends a NOTIFY message to

the calling terminal, indicating that the called party is called. At the same time, the calling

terminal plays the ringback tone to the calling party.

When the called party answers the call, the called terminal sends a CONNECT message to the

calling terminal. When receiving this message, the calling terminal stops playing the ringback

tone and the link is established on the calling party side. The called ET sends a CONNECT

ACK message.

When a party hangs up the phone, connection termination operations are performed on the

ISDN. A DISCONNECT message indicates that the connection is to be terminated. The

RELEASE and RELEASE complete messages are sent after the DISCONNECT message.





27

4.3.3 IP PBX Access to the PSTN

A PBX can interoperate with a PSTN network or another PBX through an E1 port or a PRI

according to the Q.931 protocol to interconnect an intranet with a public network.

Figure 4-5 shows the implementation.

Figure 4-5 Implementation of E1/PRI relay features

PBX IPPBX PSTN

SETUP

SETUP

ACKNOWLEDGE

Number analysis

succeeds.

CONNECT

CALL PROCEEDING

ALERTING

CALL PROCEEDING

Session is

set up.

CONNECT ACKNOWLEDGE

SETUP

ALERTING

CONNECT

CONNECT ACKNOWLEDGE

1. User A (a downstream PBX user) picks up the phone and dials the number of user B (a

PSTN user). Through the E1/PRI port, the IP PBX receives a SETUP message that

contains the number of user B.

2. The IP PBX analyzes the number of user B, and selects the E1/PRI port as the port for

sending a SETUP message (containing the number of user B) to the PSTN.





28

3. After successfully analyzing the number, the PSTN sends a CALL PROCEEDING

message to the IP PBX, indicating that the number of user B is successfully analyzed.

Then, a call is set up between the PSTN and the IP PBX.

4. The IP PBX sends a CALL PROCEEDING message to the downstream PBX, indicating

that a call is being set up.

5. The PSTN sends an ALERTING message to the IP PBX, indicating that the phone of

user B starts to ring.

6. The IP PBX sends an ALERTING message to the downstream PBX. Then, user A hears

the RBT.

7. User B picks up the phone. Then, the PSTN sends a CONNECT message to the IP PBX,

indicating that user B (the called party) has accepted the call.

8. The IP PBX sends a CONNECT message to the downstream PBX. Then, user A stops

hearing the RBT.

9. The downstream PBX sends a CONNECT ACKNOWLEDGE message to the IP PBX,

indicating that user A (the calling party) answers the call.

10. The IP PBX sends a CONNECT ACKNOWLEDGE message to the PSTN, indicating

that user A and user B are engaged in the call.

4.4 SIP UE Access

In SIP UE access, a software terminal (SIP UE) using SIP accesses the IP PBX and

registers with the IP PBX through the IP network, and uses the services provided by the IP

PBX.

A SIP UE must register with the SIP proxy server when initiating the first call. A SIP

registration process consists of three stages: SIP UE registration, re-registration, and

deregistration. The SIP UE can be registered with or without authentication. Chapter 3 "SIP"

describes the VoIP registration process. This section describes only the registration

process with authentication.





29

4.4.1 SIP UE Registration

Figure 4-6 shows the implementation of SIP UE registration.

Figure 4-6 Implementation of SIP UE registration

SIPUE IPPBX

Register(PrivateId)

Register-401(WWW-Authenticate)

Registration

succeeds

Register(PrivateId, Authorization)

Register-200(PublicId)

1. The SIP UE sends a Register message (containing the registration account) to the IP

PBX to initiate registration.

2. The IP PBX finds that the Register message does not contain the authentication

information, so the IP PBX sends a Register-401 message. The Register-401 message

contains the WWW-Authenticate header.

3. The SIP UE sends a Register message containing the registration account and the

Authorization header to the IP PBX again to initiate registration.

4. The IP PBX checks the authentication information of the SIP UE and sends a

Register-200 message after authentication is successful. The Register-200 message

contains the PublicId to be used by the SIP UE.





30

4.4.2 Process of a Call Between SIP UEs

Figure 4-7 shows the process of a call between two SIP UEs.

Figure 4-7 Process of a call between two SIP UEs

SIP UE (calling party) PBX

INVITE

INVITE-180

SIP UE (called party)

INVITE

INVITE-180

The called party hears

the ring tone.

The calling party hears the ringback tone.

The called party picks up

the phone.

INVITE-200

INVITE-200

INVITE-ACK

INVITE-ACK

Session is set up.

The calling party hangs

up the phone.

BYE

BYE


the busy tone.

BYE-200

BYE-200

The called party hangs

up the phone.





31

1. The calling SIP UE sends an INVITE message containing the called number to the IP

PBX.

2. The IP PBX locates the called SIP UE and sends an INVITE message to this SIP UE.

3. The called SIP UE rings and sends an INVITE-180 response message to the IP PBX.

4. The IP PBX forwards the INVITE-180 response message of the called SIP UE to the

calling SIP UE. Then, the calling party hears the RBT.

5. The called party picks up the phone, and the called SIP UE sends an INVITE-200

response message to the IP PBX.

6. The IP PBX forwards the INVITE-200 response message of the called SIP UE to the

calling SIP UE.

7. The calling SIP UE sends an INVITE-ACK message to the IP PBX.

8. The IP PBX forwards the INVITE-ACK message to the called SIP UE.

9. The calling and called parties start the conversation.

10. The calling party hangs up the phone, and the calling SIP UE sends a BYE message to

the IP PBX.

11. The IP PBX forwards the BYE message to the called SIP UE, and the called party hears

the busy tone.

12. The called SIP UE sends a BYE-200 message to the IP PBX.

13. The IP PBX forwards the BYE-200 message to the calling SIP UE.

14. The called party hangs up the phone and the call is ended.

4.5 Access to the IMS Through SIP

A user can access the IMS through SIP with registration or without registration.

Access without registration is easy and commonly used. This section describes only access to

the IMS through SIP with registration.





32

4.5.1 Access to the IMS Through SIP with Registration

Figure 4-8 shows the process of registration with the IMS through SIP.

Figure 4-8 Registration with the IMS through SIP

IPPBX IMS

Register(PrivateId)

Register-401(WWW-Authenticate)

Registration

succeeds

Register(PrivateId, Authorization)

Register-200(PublicId)

1. The IP PBX sends a registration message containing the registration account to the IMS

to initiate registration.

2. The IMS finds that the Register message does not contain the authentication information,

so the IMS sends a Register-401 message containing the WWW-Authenticate header.

3. The IP PBX sends a Register message containing the registration account and the

Authorization header to the IMS again to initiate registration.

4. The IMS checks the authentication information of the IP PBX and sends a Register-200

message after authentication is successful. The Register-200 message contains the

PublicId to be used by the IP PBX.





33

4.5.2 Calling and Called Parties Access to the IMS Through SIP

Figure 4-9 shows the call process in the IMS where SIP is used.

Figure 4-9 Call process in the IMS where SIP is used

IPPBX IMS

INVITE

INVITE-180

Called party (SIP

UE)

INVITE

INVITE-180

The called party hears the

ring tone.


the phone.

INVITE-200

INVITE-200

INVITE-ACK

INVITE-ACK

Session is set up.

Calling party

(POTS)

Pick up the phone

Play the dial tone

Dial number (digits)


Stop playing the ringback tone

1. The calling party picks up the phone and hears the dial tone played by the IP PBX.

2. The calling party dials the called number. The IP PBX collects all the digits and analyzes

the digits. According to the configuration, the IP PBX identifies the call to be destined

for the IMS network. Then, the IP PBX sends an INVITE message containing the called

number to the IMS.

3. The IMS locates the called party (a SIP UE) and sends an INVITE message to the called

party.

4. The called SIP UE rings and sends an INVITE-180 message to the IMS.

5. The IMS forwards the INVITE-180 message of the called SIP UE to the IP PBX. Then,

the IP PBX plays the RBT to the calling party.

6. The called party picks up the phone, and the called SIP UE sends an INVITE-200

message to the IMS.

7. The IMS forwards the INVITE-200 message of the called SIP UE to the IP PBX.

8. The IP PBX sends an INVITE-ACK message to the IMS.





34

9. The IP PMS forwards the INVITE-ACK message to the called SIP UE.


4.6 PBX Communication Through SIP

Figure 4-10 shows the call process between the IP PBXs connected through SIP.

Figure 4-10 Call process between IP PBXs connected through SIP

Calling party IP

PBX

Called party IP

PBX

INVITE

INVITE-180

Called party (SIP

UE)

INVITE

INVITE-180


the ring tone.

INVITE-200

INVITE-200

INVITE-ACK

INVITE-ACK

Session is set up.

Calling party

(POTS)

Pick up the phone

Play the dial tone

Dial number (digits)


Stop playing the ringback tone


the phone.

1. The calling party picks up the phone and hears the dial tone played by the calling IP

PBX.

2. The calling party dials the called number. The calling IP PBX collects all the digits and

analyzes the digits. According to the configuration, the calling IP PBX identifies that the

destination of this call is another IP PBX (the called IP PBX). Then, the calling IP PBX

sends an INVITE message containing the called number to the called IP PBX.

3. The called IP PBX locates the called party (a SIP UE) and sends an INVITE message to

this SIP UE.

4. The SIP UE of the called party rings and sends an INVITE-180 message to the called IP

PBX.





35

5. The called IP PBX forwards the INVITE-180 message of the called party to the calling

IP PBX, and then the calling IP PBX plays the RBT to the calling party.

6. The called party picks up the phone, and the SIP UE of the called party sends an

INVITE-200 message to the called IP PBX.

7. The called IP PBX forwards the INVITE-200 message of the called party to the calling

IP PBX.

8. The calling IP PBX sends an INVITE-ACK message to the called IP PBX.

9. The called IP PBX forwards the INVITE-ACK message to the called party.


4.7 Fax/Modem

With the fax service, the IP PBX carries data transmitted from fax machines on both sides of a

network and manages services. SIP UE and POTS terminals are supported. Data can be

transmitted through the SIP relay or between local users.

Auto-Switch

User A is a POTS user, and user B is a SIP UE user. A call is set up between user A and user B.

The process is similar to that of a telephone call. Then, the fax machine transmits a called

terminal identification (CED) and calling tone (CNG). When an operator presses the Start

button on the fax machine on the receiver side, the CED is transmitted to the peer end,

indicating that the receiver is ready for receiving. After CNG is transmitted on the sender side,

indicating that the sender is ready for transmitting, the voice channel is blocked, and voice

communication cannot proceed. After CED and CNG are detected, A and B automatically

switch to the corresponding codec (T.38, G.711, or CLEARMODE) based on the

configuration, and automatically switch to voice mode when the fax completes.

Locally Initiated Negotiation

User A is a POTS user, and user B is a SIP UE user. User A and user B set up a call

connection. Then they detect CED and CNG and disable the voice channel. User A sends a

Reinvite message to the SIP UE for codec negotiation based on the configuration. When

detecting the fax completion, user A sends an Offer message for negotiation. After the

negotiation, user A switches to voice mode.

Remotely Initiated Negotiation

User A is a POTS user, and user B is a SIP UE user. User A and user B set up a call

connection. The SIP UE detects a fax signal and initiates a negotiation. User A receives the

negotiation message and negotiates codec based on the configuration. When detecting the fax

completion, user A sends an Offer message for negotiation. After the negotiation, user A

switches to voice mode.

4.8 Number Change

By customizing rules, you can change numbers and call prefixes to implement the number

change service. Second number analysis is performed and the second dial tone is played after

a number changes.





36

Number Change Type

Numbers can be changed in following modes:

Calling number screening: The calling number is changed when a user initiates a call.

Pre-routing number change: The calling/called number is changed after number analysis

and before route selection. Second number analysis is performed and the second dial

tone is played after a number changes.

Post-routing number change: The calling/called number is changed after route selection.

Calling Number Change

It can be configured in calling number discrimination, number change before route selection,

and number change after route selection. For example, calling number discrimination can be

configured for a voice customer service center as follows: Change the numbers of all outgoing

calls with DN set 0 to 95555. Then, the numbers of all outgoing calls will be changed to

95555.

Called Number Change

It can be configured in number change before route selection and number change after route

selection. For example, the following rule of number change before route selection is

configured for a company: Change the prefix of the called number from 029 to 17909+029.

When a calling party dials 029+called number, the called number will be changed to

17909+toll call number.

DN Set Change

A dial number (DN) set defines a group of numbers that are processed in the same way. A DN

set, a country code, and an area code identify the home area of a user. A DN set and a call

prefix determine the dial plan for a user. DN sets divide a physical network or a device into

multiple logical networks.

It can be configured in calling number discrimination, number change before route selection,

and number change after route selection. The IP PBX can change the DN set during a call to

implement fast reusing of the dialing scheme.

Second Dial Tone Playing

It can be configured in number analysis before route selection. Then, the IP PBX will play the

second dial tone to the user if the user dials an outgoing call prefix.

4.9 Intelligent Routing

Routes are selected based on the user-defined rules. AR routers can select routes based on

5-tuple information (DN set, Centrex group ID, calling number, call prefix, and time range).

Selecting Routes Based on Time Ranges

For example, when the IP PBX is configured with dual upstream routes (route A is SIP IP

trunk and route B is PRA trunk), outgoing calls are directed to route A from 8:00 a.m. to 6:00

p.m. and are directed to route B during other time ranges.





37

Selecting Routes Based on User Types

For example, when the IP PBX is configured with dual upstream routes (route A is SIP IP

trunk, and route B is PRA trunk where the voice quality is better), the IP PBX can select route

B for customer groups with a higher priority.





38

4.10 CDR

The call detail record (CDR) of users can be queried in real time, and the CDR data can be

analyzed by using a third-party tool. In this way, users can quickly learn the fee of a call in

process and the total fee of the entire call.

Figure 4-11 shows the principle.

Figure 4-11 CDR principle

(SIPUE)

User AIPPBX

(SIPUE)

User B

Invite

180

Invite

180

200 OK

200 OK

ACKACK

Record

CDR startBye Bye

200 OK200 OK

Record

CDR end

1. Set the FTP server address by using a command.

2. User A dials the number of user B. User B picks up the phone and a call is set up.

3. The IP PBX records the CDR start information, including the call start time, calling party

information, and called party information.

4. If either user hangs up the phone to terminate the call, the IP PBX records the CDR end

information and sends the CDR data to the specified FTP server.


Voice Feature White Paper 5 IVR Service



39

5 IVR Service

The Interactive Voice Response (IVR) service allows enterprises to customize their IVR menu

and prompt tone, improving user experiences.

When there is an incoming call to the access code of an IVR service, the user is prompted to

dial the extension number or the exchange number. The user can dial the extension number to

enter the start conversation with the called party or dial the exchange number to trigger the

simultaneous ringing service, sequential ringing service, line selection service (selecting a

user based on certain rules). When multiple users call the IVR service, subsequent users wait

in the call queue and hear the call queuing announcement.

5.1 Dialing an Extension Number

An IP PBX can trigger IVR automatic connection. When there is an incoming call to the

access code of an IVR service, the IP PBX connects the call, and the user is prompted to dial

the extension number or the exchange number. This section describes the basic calling process

in which the user dials the extension number.

Assume that user A calls user B by dialing the access code of an IVR service, and users A and

B use SIP UEs. The process is shown in Figure 5-1.





40

Figure 5-1 Process of dialing an extension using the IVR

16.200(SDPD)

1.INVITE(SDPA)

2. Req_uri is the access code of an IVR service, triggering

the IVR service.

4.ACK

8.INVITE(SDP_ringback tone)

9.200(SDP)

14.200(SDPA)

6.INVITE(SDP)

7.180

10.ACK11.200(SDP)

18.ACK

Session is set up.

User A User BIP PBX

3.200(SDP_play tone)

5. User A hears the IVR prompt tone and calls user B.

12.ACK13.reINVITE

15.reINVITE(SDPA)

17.ACK(SDPD)

1. User A dials the access code of an IVR service.

2. The IP PBX checks the RequireURI of user A and triggers an IVR service if the

RequireURI is the access code of the IVR service.

3. The IP PBX returns message 200 and plays the IVR prompt tone to user A.

4. User A sends ACK.

5. User A hears the IVR prompt tone and dials the number of user B.

6. When receiving the number of user B dialed by user A, the IP PBX initiates a call to user

B.

7. User B returns message 180.

8. The IP PBX sends Invite to user A.

9. User A returns message 200.

10. The IP PBX returns ACK and plays the ringback tone to user A.

11. User B picks up the phone and returns message 200.

12. The IP PBX returns ACK to user B.





41

13. The IP PBX sends Reinvite to user A without SDP.

14. User A returns message 200 with SDP.

15. The IP PBX sends Reinvite to user B with the SDP of user A.

16. User B returns message 200 with SDP.

17. The IP PBX returns ACK to user A with the SDP of user B.

18. The IP PBX returns ACK to user B. The session is set up.

5.2 Triggering a Simultaneous Ringing Service


access code of an IVR service, the user is prompted to dial the extension number or the

exchange number. This section describes the process of triggering the simultaneous ringing

service. When a user dials the exchange number, the simultaneous ringing service is triggered,

and all idle phones ring in the group. When a phone is picked up to answer the call, other

phones stop ringing.

Assume that an exchange group consists of users B, C, and D. When user A calls the access

code of an IVR service, the process is shown in Figure 5-2.





42

Figure 5-2 Process of triggering the simultaneous ringing service by dialing the exchange number

20.ACK(SDPD)

1.INVITE(SDPA)

Req_uri is the access code of an IVR service,

triggering the IVR service.

3.ACK

8. reINVITE(SDP_play tone)

9.200(SDPA)

16.200(SDPA)

4.INVITE(SDPA)

7.180

10.ACK

11.180

21.CANCEL

User B is ringing

User A User BIPPBX

2. 200(SDP_play tone)

User A hears the IVR prompt tone and dials the exchange number.

14.ACK

12.180

17.RE-INVITE(SDPA)

15.RE-INVITE

User C User D

5.INVITE(SDPA)

6.INVITE(SDPA)

User C is ringing

User A is listening ringback tone.

User D is ringing

13.200(SDPD)

18.200(SDPD)

19.ACK

User A is talking with User D.

22.200

23.487

24.ACK

User B stop ringing

25.CANCEL

26.200

27.487

28.ACKUser C stops ringing.



RequireURI is the access code of the IVR service. The IP PBX returns message 200 and

plays the IVR prompt tone to user A.

3. User A sends ACK. User A hears the IVR prompt tone and dials the exchange number.

4. The IP PBX sends Invite to user B.

5. The IP PBX sends Invite to user C.





43

6. The IP PBX sends Invite to user D.

7. User C rings and returns message 180.

8. The IP PBX sends Reinvite to user A.



11. User B rings and returns message180.

12. User D rings and returns message 180.

13. User D picks up the phone and returns message 200.

14. The IP PBX returns ACK to user D.


16. User A returns message 200 with SDP.

17. The IP PBX sends Reinvite to user D with the SDP of user A.

18. User D returns message 200 with the SDP of itself.


20. The IP PBX returns ACK to user A with the SDP of user D. The session is set up

between users A and D.

21. The IP PBX sends Cancel to user B to cancel the invite request. When receiving the

message, user B stops ringing.

22. User B returns Cancel-200.

23. User B returns Invite-487.

24. The IP PBX returns an ACK response message to user B.

25. The IP PBX sends Cancel to user C to cancel the invite request. When receiving the

message, user C stops ringing.

26. User C returns Cancel-200.

27. User C returns Invite-487.

28. The IP PBX returns an ACK response message to user C.

5.3 Triggering a Sequential Ringing Service



exchange number. This section describes the process of triggering the sequential ringing

service. After the sequential ringing service is triggered, all idle phones in the group ring in

sequence. When ringing timeout occurs on the previous phone, the next phone rings. When a

phone is picked up and the call is set up, subsequent phones do not ring.

Assume that the exchange group consists of users B, C, D, and E. Users B and D use SIP UEs

and users C and E use POTS phones. When user A dials the access code of an IVR service,

the process is shown in Figure 5-3.





44

Figure 5-3 Process of triggering the sequential ringing service by dialing the exchange number

27.ACK(SDPD)

1.INVITE(SDPA)


the IVR service.

4.ACK

8. reINVITE(SDP_play tone)

9.200(SDPA)

6.INVITE(SDPA)

10.ACK

13.200

12.CANCEL

User B stop ringing

User A User BIPPBX


5. User A hears the IVR prompt tone and dials the exchange number.

16.Ring

18.StopRing

23.RE-INVITE

User C User D

22.ACK

User A is listening ringback tone

20.180

21.200(SDPD)

19.INVITE(SDPA)

24.200(SDPA)25.RE-INVITE(SDPA)

26.200(SDPD)

28.ACK

User A is talking with User D

7.180

User B is ringing

11. The sequential ringing timer expires and

the call is released.

14.487

15.ACK

User C is ringing

17. The sequential ringing timer expires and

the call is released.

User C stop ringing



2. The IVR service is triggered.


4. User A sends ACK.

5. User A hears the IVR prompt tone and dials the exchange number.


7. User B rings and returns message180.






45



11. When the sequential ringing timer expires, the call to user B is released.

12. The IP PBX sends Cancel to user B. User B stops ringing.

13. User B returns Cancel-200.

14. User B returns Invite-487.

15. The IP PBX returns ACK.

16. The IP PBX sends Ring to user C. User C rings.

17. When the sequential ringing timer expires, the call to user C is released.

18. The IP PBX sends StopRing to user C. When receiving the message, user C stops

ringing.


20. User D rings and returns message 180.




24. User A returns message 200 with the SDP of itself.

25. The IP PBX sends Reinvite to user D with the SDP of user A.


27. The IP PBX returns ACK to user A with the SDP of user D.

28. The IP PBX returns ACK to user D. The session is set up between users A and D.

5.4 Triggering the Line Selection Service



exchange number. This section describes the process of triggering the line selection service.

When the line selection service is triggered, the IP PBX selects a user from all idle users and

plays the ring tone based on configured line selection rules (ascending, descending, or polling

order of user index numbers). If a user picks up the phone and answers the call, the call is set

up. If the ringing times out, the call is stopped and subsequent users do not ring.

Assume that the exchange group consists of users B, C, D, and E. User B uses SIP UE.

According to the IVR line selection rule, user B is called. When user A dials the access code

of an IVR service, the process is shown in Figure 5-4.





46

Figure 5-4 Process of triggering the line selection service

16.200(SDPD)

1.INVITE(SDPA)


the IVR service.

4.ACK

8. INVITE(SDP_ringback tone)

9.200(SDP)

14.200(SDPA)

6.INVITE(SDP)

7.180

10.ACK11.200(SDP)

18.ACK

The session is set up.

User A User BIPPBX


User A hears the IVR prompt tone and dials the exchange number.

12.ACK13.reINVITE

15.reINVITE(SDPA)

17.ACK(SDPD)

5. A call is initiated to user B based on the line

selection rule.





4. User A sends ACK. User A hears the IVR prompt tone and dials the exchange number.

5. The IP PBX selects idle user B to receive the call based on the line selection rule.


7. User B rings and returns message 180.




11. User B picks up the phone and returns message 200.

12. The IP PBX sends ACK to user B. User B stops ringing.






47

14. User A returns message 200 with the SDP of itself.

15. The IP PBX sends Reinvite to user B with the SDP of user A.

16. User B returns message 200 with the SDP of itself.

17. The IP PBX returns ACK to user A with the SDP of user B.

18. The IP PBX returns ACK to user B. The session is set up between users A and B.

5.5 Triggering Call Queuing



exchange number. If a user calls when all exchange users are busy, call queuing is triggered.

While waiting, the calling party hears the IVR queuing prompt tone. When an exchange user

is idle, the call is set up.

Assume that an exchange group consists of users D, E, and F. Users A, B, and C are

talking with D, E, and F respectively. When user G calls the IVR exchange, call queuing is

triggered. The process is shown in Figure 5-5.





48

Figure 5-5 Process of triggering the call queuing

19.ACK

1.INVITE(SDPA)

6. RE-INVITE(SDP_play tone)

7.200(SDPA)

14.200

4.ACK

8.ACK

11.BYE

23.200(SDPA)

User G User BUser A

User A is talking with User D

13.200

16.180

User C IPPBX

5. User G hears the IVR prompt tone and dials the exchange number.

12.BYE

17. INVITE(SDP_ringback tone)

18.200(SDP)

20.200(SDPD)

22.RE-INVITE21.ACK

The session is set up between users D and G.

User D User E User F

User B is talking with User E

User C is talking with User F

2. Req_uri is the access code of an IVR service, triggering the IVR

service.


9. User G is listening the IVR queuing prompt tone.

10. User D hangs up the phone and the session between users

A and D is released.

15.INVITE(SDPA)

24.RE-INVITE(SDPA)

25.200(SDPA)

26.ACK(SDPD)

27.ACK

1. User G dials the access code of an IVR service.

2. The IP PBX checks the RequireURI of user G and triggers an IVR service if the


3. The IP PBX returns message 200 and plays the IVR prompt tone to user G.

4. User G sends ACK.

5. User G hears the IVR prompt tone and dials the exchange number.

6. The IP PBX determines that all exchange users are busy, makes user G to wait in a queue,

and sends Reinvite to user G.

7. User G returns message 200.

8. The IP PBX returns ACK.

9. User G hears the IVR queuing prompt tone.

10. User D hangs up the phone and releases the call with user A.





49

11. User D sends Bye to the IP PBX.

12. The IP PBX sends Bye to user A.


14. The IP PBX returns message 200 to user D.


16. User D returns message 180.

17. The IP PBX sends Reinvite to user G.

18. User G returns message 200.

19. The IP PBX returns ACK and plays the ringback tone to user G.



22. The IP PBX sends Reinvite to user G without SDP.

23. User G returns message 200 with the SDP of itself.

24. The IP PBX sends Reinvite to user D with the SDP of user G.


26. The IP PBX returns ACK to user G with the SDP of user D.


User G is talking with exchange member D.


Voice Feature White Paper 6 BEST Function Description



50

6 BEST Function Description

6.1 Overview

In centralized call control deployment mode, remote branches fail to provide cost-effective

backup capability when enterprises deploy IP telephones and high-value applications from the

central site to remote branches. Most enterprise cannot professional call processing server and

unified information processing server or provide multiple WAN links at all remote branches

due to their quantity and scale. In Huawei IP communication solution, HW Call Manager (CM)

is used together with Huawei Branch Exchange for Survivable Telephony (BEST) feature

provided by VSP software, enabling enterprises to deploy high-availability IP telephone at

branches.

6.2 Description

Figure 6-1 BEST deployment

In normal operation mode, branches are connected to the CM (the AR serves as the CM) of

the headquarters, and provide call processing services using the IP WAN. When the IP WAN


Voice Feature White Paper 6 BEST Function Description



51

link of the CM is disconnected or both the active and standby CMs break down, the remote

BEST voice regeneration function of the local voice gateway on the branch must be enabled

to maintain local voice communication, including the registration and call management of

local POTS telephone and IP telephone. In addition, branches provide call route backup to the

PSTN using the backup AT0/BRI trunk. The BRI trunk is supported in AR V200R002C02.

When the IP WAN link resumes or the CMs recover, the VoIP services are automatically

switched back to the CM in the headquarters, and the POTS calls on the CMs in the

headquarters are resumed using the SIP AG.


Voice Feature White Paper 7 Power Outage Survival



52

7 Power Outage Survival

When the voice gateway is powered off or fails to work properly, the power outage survival

function can ensure the connection of important calls, and improve the reliability and

availability of enterprise voice calls.

Figure 7-1 Power outage survival deployment

In VoIP applications such as the IP PBX and IP call center, port power outage survival is an

ultimate backup solution to the local system. The 4FXS1FXO card of Huawei AR G3 series

router provides one port for survival. When the AR G3 telephone system stops working, the

call can be directly connected to the PSTN line.

As shown in Figure 7-1, the FXO port is used to connect to the PSTN, and the FXS port is

used to connect to the analog telephone set. During the normal communication, all outgoing

IP calls and analog calls can be connected to the local PSTN over the FXO port. When the

router in AR G3 series is powered off due to accidents, the IP calls cannot be connected.

However, one 4FXS1FXO card can connect the FXO line to the nearest FXS interface,

because only the FXS interface can be used for power outage survival. In this way, a channel

of analog calls is reversed for the IP PBX to ensure normal communication.


Voice Feature White Paper 8 Call Manager System



53

8 Call Manager System

The CM system provides enhanced functions for AR G3 series routers, and the PBX or hybrid

PBX functions for enterprises. With the CM system, enterprises can deploy data connections

and call solution on their existing networks. The CM system provides advanced functions that

cannot be implemented in multiple call solutions. It combines IP call and data routing service

in a single solution, meeting customers' service requirements at lower O&M costs.

8.1 Advantages

The VoIP technology develops rapidly and its cost advantage has been recognized by

enterprises as it integrates the voice, video, and data services on a single network. Because the

CM system is integrated on a router, the system features the following specific advantages:

Single integrated voice and data platform providing cost-effective operation for all

enterprise branches: The AR G3 series routers provide functions such as the QoS,

network security, encryption, and firewall for enterprise offices. The routers provide

integrated functions, such as IP call, voice message, and automatic answer. One device

can meet all the service requirements, simplifying management, operation, and

maintenance, and reducing the total cost of ownership.

Unified key system and IP PBX function: Specific functions must be provided for

different enterprises. The CM system provides powerful value-added voice features to

improve the production efficiency of end users and enterprises.

Scalability: The AR G3 series routers provide the CM system with flexible interfaces

such as the PSTN interfaces and widely-used WAN interfaces. These interfaces improve

the flexibility of the CM system, and facilitate the integration of voice services and data

services.

Items supported by the CM system are as follows:

User management and call management of the POTS and ISDN telephones, and the SIP

UE

All PBX services

Various trunks such as the AT0, E1, and SIP


Voice Feature White Paper 8 Call Manager System



54

8.2 Deployment

Figure 8-1 CM deployment

VOICE

VOICE

SIP AG

IP PBX

Call Manager

POTS user FAX PC userSIP user

VOICE

IP NetwortPSTN

POTS user

FAX

PC userSIP user

SIP AG

Headquarters

Branch


Voice Feature White Paper 9 Other Services Supported by AR G3 Series Routers



55

9 Other Services Supported by AR G3 Series Routers

In addition to basic IP PBX services, the AR G3 series routers provide supplementary IP PBX

services. Customers can enable or disable the supplementary IP PBX services according to

their own needs.

Table 9-1 lists other supported services when AR G3 series routers function as the IP PBXs.

Table 9-1 Other supported services when AR G3 series routers function as the IP PBXs

Service Type Description

Calling line

identification

presentation

(CLIP) service

Generally, the IP PBX sends the calling number to the called party, and

displays the calling number on the called telephone or corresponding

terminal device. If the calling party has the calling line identification

restriction service, the calling number cannot be displayed on the called

terminal device.

Calling line

identification

restriction

(CLIR) service

Restricts the display of the calling number to the called party during the

establishment of a call.

Calling Line

Identification

Restriction

Override (RIO)

service

If the called party has the RIO service, the calling number information can

still be displayed on the called party, even though the calling party has the

CLIR service.

Temporarily

activating the

CLIR

restriction

Normally, the calling number is displayed on the called phone. By dialing

a certain prefix before the called number, the called party can prevent the

calling number from being displayed on the called phone in this call.

Temporarily

canceling the

CLIR service

Normally, the calling number is not displayed on the called phone. If the

calling party dials a certain prefix before the called number, the calling

number can be displayed on the called phone in this call.





56


Call hold During the conversation between user A (service user) and user B, user A

can press hookflash to suspend the call, and handle a temporary

emergency. In this case, user A can hear a special dial tone, and user B can

hear a hold tone (for example, a piece of music). User A can press

hookflash again to restore the call. The call hold service is the basis of

other hooking services.

Double

communication

(DC)

When the call between user A and user B enters the call hold status, user A

can place a call to user C, and switch between the two conversations.

Call waiting When user C places a call to user A who is in conversation, user A can

allow user C to wait. Meanwhile, the system displays a number or plays a

prompt tone to tell user A that a subscriber is waiting. User A can release

the ongoing call or hold the call and then connect to user C.

Call transfer

service

User B (transferor) who is in conversation with user A (transferee) can

transfer the call to user C (transfer target). User B releases the call, and the

call between user A and user B is set up. With the call transfer service, the

service user can transfer the ongoing call to a third-party user by

performing corresponding operating.

Three-party

service

After the call between user A (service user) and user B is set up, user A

places a call to user C, and then allows user B and user C to join the

conversation. Then, a three-party voice conversation is set up.

Call forwarding When a user is called, the call is forwarded to a preset party if the user

registers the call forwarding service and the call flow satisfies forwarding

conditions.

Call forwarding-unconditional (CFU): All calls of a service user are

forwarded to a preset party unconditionally. The AR supports remote

registration of the CFU service.

Call forwarding-busy (CFB): When a user places a call to a service

user who is in conversation, the call is forwarded to a preset party. The AR

supports remote registration of the CFB service.

Call forwarding-no reply (CFNR): When a call placed to a service user is

not answered, the call is forwarded to a preset party. The AR supports

remote registration of the CFNR service.

Call forwarding-offline (CFO): When the called user is offline, the

incoming call is forwarded to the preset party. (Offline: The IP PBX

changes the user status to the offline status if new subscription of the user

is not refreshed after subscription timeout.) The AR supports remote

registration of the CFO service.

Completion of

Calls to Busy

Subscriber

(CCBS)

The IP PBX monitors the called party status when the called party is busy.

When the called party is idle, the IP PBX notifies the calling party so that

the calling party can determine whether to make a call to the called party

again.

Completion of

Communicatio

n on no Reply

(CCNR)

When the called number is busy, the IP PBX monitors the called party

status. When the called party is idle, the IP PBX notifies the calling party,

and determines whether to make a call according to the status of the

calling and called parties.





57


Callback A service user can dial the number of the last incoming call using the

service code.

Missed call A service user can hear the voice prompt of the last missed call, and dial

the number of the last missed incoming call.

Redial A service user can redial the number of the last outgoing call.

Password call

service

A service user can use or limit the international toll of the telephone by

setting the password. The password call priority level is higher than other

call-out rights. Even if a user has no call-out right, the user can place a call

using the password call service.

Password call

barring

A service user can use or limit the inter-office calls of the telephone by

setting the password.

Number barring Call initiation is limited. When a user registering this service calls by

dialing the preset barred number, the call is barred.

Direct dialing

to access the

system

An external user can call an internal by dialing the PSTN long number of

the internal user. The call does not need to be transferred to this user using

the automatic switchboard.

Do-not-disturb A user who does not expect to be called can use this service. After the

do-not-disturb service is registered, all incoming calls to the user are

answered by the IP PBX. Outgoing calls of the user are not affected. The

special dial tone is played after the calling user picks up the phone.

Reject

anonymous call

The service allows a service user to reject an anonymous call (with no

calling number displayed) and allows the system to play an announcement

to the calling number after call rejection.

Selective call

acceptance

If an incoming call meets the requirements preset by the service user, the

call is connected. If not, the call is barred.

Selective call

rejection

If an incoming call meets the requirements preset by the service user, the

call is barred. If not, the call is connected.

Simultaneous

ringing

When a calling party dials the access number of a simultaneous ringing

group, all member phones in the group ring simultaneously, and the called

party can answer the call using any ringing phone.

Sequential

ringing

When a calling party dials the access number of a sequential ringing

group, member phones in the group ring in the configured sequence.

Co-group

pickup

Users A and B belong to the same pickup group. When user B's phone

rings, user A (service user) answers a call addressed to user B by dialing

the pickup access code.

Designated

pickup

When the phone of user A rings, user B in the same group can dial the

service access code plus user A's phone number to answer the call.

One number

link you

(ONLY)

Multiple terminals of the service user share one number, and the

sequential ringing and simultaneous ringing are supported.





58


PBX line

selection

The system selects a called party from the group based on the preset

selection mode when an outer-group user calls the primary number of the

PBX group. Lines can be selected based on the index numbers of users in

a group in ascending, descending, or polling order.

Short number

calling service

Calls between users are placed by dialing short numbers.

Call

interception

When a call placed by a service user fails, the system switches the call to

the IVR system. The IVR system plays a user-friendly tone to guide user's

operating.

Local number

querying

service

After a service user dials the service opcode, the IP PBX can announce the

number of the user in voice.

Distinctive

ringing service

Delivers the different ringing to users (including local, national,

international, and intergroup users) based on prefix analysis.

Abbreviated

dialing

The 2-digit abbreviated code is dialed instead of the original called

number. That is, the user can make a call by dialing the 2-digit abbreviated

code, without having to dial the original called number.

Wake-up

service

If a service user sets a wake-up request, the system sends the wake-up tone

to the user when the preset time is up.

Multiple user

number

A user can be connected by dialing the standby or active number of the

user. An active number can correspond to multiple standby numbers, and

a standby number can correspond only to one active number.

Ring back tone When a call to a service user is placed and is not answered, the calling user

can hear the customized ring back tone.

Remote office

service

The remote office service allows a user to access from any terminal and

share original services such as short number dialing and call transfer.

Secretary Allows a user to designate another phone number (for example, the

secretary's phone number) to process all incoming calls. All incoming

calls of the user are transferred to the secretary's phone number first, and

only the secretary can call the user directly.

Call resident

service

This call parking service allows a user to place an ongoing call on hold

and then retrieve a call that is placed on hold on another phone within the

same group. If the call is not retrieved on another phone within the

specified period, the service user's phone rings.

SCC

deregistration

A service user can cancel all supplement services by dialing the SCC

number.


Voice Feature White Paper 10 SIP NAT Traversal



59

10 SIP NAT Traversal

10.1 Overview

The VPN access function is unavailable in most small- or medium-sized enterprise. The

employees on business travel may work on a private network. When IP PBX services are used

in these enterprises, SIP UE subscribers who roam into another place can access the VoIP

system of the headquarters only if SIP and H.323 signaling messages can traverse the firewall.

The reason is that: On the private network, a signaling address is carried by a packet, while

the media stream address is negotiated dynamically using a signaling protocol. The signaling

address and media address are private IP addresses, and cannot be routed on the public

network.

An enterprise can also access a telecom operator's network using a VoIP trunk. The IP address

used to access the network may be different from the IP address of a SIP UE in the enterprise.

This chapter describes how to ensure the interaction of VoIP services between the SIP UE on

the intranet and extranet.

Figure 10-1 SIP network address translation (NAT) traversal

SIP Server

AR Enterprise

egress routerPublic network

Private network Signal:10.138.1.3

Media:10.138.1.3

202.38.64.10

SIP UE

IMS

SIP relay (public IP)

202.10.88.5

10.138.2.5

SIP





60

10.2 SIP NAT Traversal Principles

The payload of a SIP message carries address information. For example, the Contract field

describes the signaling contact address, and the SDP field describes the media address.

However, the NAT function translates only the addresses in the UDP/TCP packet header and

IP packet header, and does not translate the address in payload. As a result, the IP addresses in

the IP packet header and UDP/TCP packet header are translated into public IP addresses, but

the IP address in payload is still a private IP address. Therefore, signaling and media

connections that require the address in the payload cannot be set up.

The following technologies can be used to implement NAT traversal for SIP signaling and

media streams:

1. NAT ALG: The NAT application level gateway (ALG) can identify application layer

protocols on the NAT device, and translate the addresses in protocol packets to a public

address. However, the deployment mode is difficult to implement because all NAT

devices must be upgraded, and these devices must be upgraded again when a new

application protocol is added. For SIP ALG call processes, see section 3.5 "VoIP (SIP)

MT Process" of RFC3665.

2. Middlebox communication (MIDCOM) for RFC3303: In the MIDCOM architecture,

Middlebox (NAT/FW) is controlled by using the trustable MIDCOM Agent. VoIP

protocols are identified by the MIDCOM Agent instead of the Middlebox. The

MIDCOM Agent is integrated on a call control server, such as SIP server. Therefore,

VoIP protocols are transparent to the Middlebox.

3. Proxy technologies: In the IMS solution, a session boarder controller (SBC) is used to

solve the SIP traversal problem in the following way: The SBC reassigns the address and

port for receiving signaling and RTP streams from the intranet or extranet. In this case,

the NAT between different network areas, such as between the public network and

private network, is implemented to support NAT traversal for signaling/media streams.

Signaling and media streams can be transmitted directionally by using proxy

technologies without special requirements for universal NAT networking devices.

Therefore, existing NAT devices do not need to be upgraded and telecom operators can

carry out services smoothly.

4. Tunnel mechanism: The tunnel-based traversal is to encapsulate data flows that need to

traverse the NAT device into tunnels so that the data flows do not need to be processed

by NAT device or firewall. Logically, the tunnel mechanism is composed of the tunnel

client and tunnel server. The tunnel client and tunnel server establish a tunnel using a

tunneling protocol, allowing signaling and media stream to transparently traverse the

NAT device. The tunnel client does not need to identify call signaling protocols, such as

H.323, SIP, MGCP or H.248. The tunnel server identifies signaling protocols and

forwards signaling messages to a server on the public network. Usually, the tunnel client

is on a private network, and the tunnel server is on a public network. With the tunnel

mechanism, NAT traversal can be implemented without upgrading existing NAT devices.

Only several known ports on the NAT devices need to be enabled by configuring

relevant policies. In addition, the tunnel traversal mechanism can easily implement

multi-level NAT traversal.

5. Simple traversal of UDP through NAT (STUN), first defined in RFC3489 and then

replaced by the RFC5389: The functions on the STUN client are integrated into the SIP

entities on a private network. Before a call is originated to the extranet, a SIP entity

sends a STUN to the STUN server to obtain the external addresses configured on the

NAT device, including signaling and media addresses. In subsequent SIP call signaling,

the SIP entity fills in the external addresses in the local address field. STUN is not

applicable to symmetric NAT and TCP NAT traversal. In addition, because RTP and

RTCP use the external ports obtained from the NAT device, the rule (RTCP port = RTP





61

port + 1) is disobeyed. As a result, the end-to-end RTCP packet transmission fails. To

solve this problem, use a=rtcp or a=rtcp-mux in the SDP describe RTCP ports. For

details, see RFC3605 or RFC5761.

6. Traversal using relay NAT (TURN) defined in RFC5766: The mode is similar to STUN.

The functions on the TURN client are integrated into the SIP entities on a private

network. Before initiating communication with an extranet, an SIP entity requests an

address and a port from the TURN server. Subsequent signaling and media packets are

forwarded using the TURN server in relay mode. The TURN mode is applicable to

symmetric NAT and TCP NAT traversal. The TURN server ensure that RTCP port =

RTP port + 1.

7. Interactive connectivity establishment (ICE) defined in RFC5245: It is not a new

protocol, but combines the STUN, TURN, and RSIP. ICE uses these protocols in

suitable cases to overcome shortcomings brought by only one protocol.

8. Adding response-port (rport) in the Via header field: The request receiver uses the

response-port (rport) parameter to record the source port when receiving a request

message, and uses the received parameter to record the source IP address. The receiver

uses the received&rport parameter to send subsequent response messages, ensuring

NAT traversal of response messages. For details, see RFC3581. The method can be used

to only implement traversal of SIP signaling response messages.

10.3 AR SIP NAT Traversal Solution (SBC Solution) Signaling proxy

The AR can serve as the IP PBX and process all registration and call messages of users.

When receiving signaling messages from external users, the AR processes signaling

messages and forwards them to the IMS (SBC). The AR is a user for the IMS (SBC). The

IMS sends a call request to the AR. The AR processes the request, and then forwards it to

the real called party.

Media proxy

All media streams that are exchanged between users in on the internal network and

external networks are processed and forwarded using the RTP/RTCP over the AR. The

AR checks packet validity, and determines media stream forwarding policies including

packet filter policy, QoS policy, and address translation policy based on the signaling

process results. Then the AR specifies the address and port for receiving RTP streams of

intranet and extranet users. In this way, media streams are forwarded properly and

guaranteed good QoS and security in any networking.





62

Figure 10-2 Processes in the SBC solution

CPE SBC

192.168.0.1 192.168.0.2 65.65.65.1 65.65.65.98

Media Proxy functon

enabled(SBC embeded)

192.168.0.1:10248

65.65.65.1:61448

INVITE

Destination Addr:65.65.65.1 5060

SDP:65.65.65.98 42812

INVITE


SDP:192.168.0.2 61440

200 OK

Source Addr:192.168.0.1 5060

SDP:192.168.0.1 10248

200 OK

Source Addr:65.65.65.1 5060

SDP:65.65.65.1 61448

RTP

Source Addr:192.168.0.1 10248


RTP

Source Addr:65.65.65.1 61448


RTP

Source Addr:65.65.65.98 42812

Destin Addr:65.65.65.1 61448

RTP

Source Addr:192.168.0.2 61440

Destin Addr:192.168.0.1 10248

65.65.65.98:42812

192.168.0.2:61440

1. The SBC sends an Invite message to the CPE. Assume that SDP carries the local media

address 65.65.65.98, and the UDP port number is 42812. After receiving the SIP

message, the SIP server/proxy on the CPE changes the SDP media address to a private

network address 192.168.0.2, and changes the UDP port number to 61440. The SIP

server/proxy then sends the Invite message to a SIP phone and records proxy mapping

entries.

2. The SIP phone receives the Invite message, records the media address and port number

192.168.0.2:61440 of the peer end, and responds with a 200 OK message. In this case,

the SDP carries the local media address 192.168.0.1, and the UDP port number is

changed to 10248.

3. The CPE receives the 200 OK message from the SIP phone, changes the media address

to the public network address 65.65.65.1, and changes the UDP port number to 61448. In

addition, the CPE records the mapping the original media address and the new media

address.

4. The message is sent from the SIP phone to the destination media address

192.168.0.2:61440. The CPE changes the destination media address to 65.65.65.98

42812 and sends the message to the SBC.

5. The SBC sends the message to the destination media address 65.65.65.1:61448. The CPE

changes the phone destination media address to 192.168.0.1:10248 and sends the

message to the SIP phone.


Voice Feature White Paper 11 AR Voice Solution



63

11 AR Voice Solution

An AR G3 software package integrates all functions of the SIP AG and IP PBX, and can meet

various networking requirements, such as voice gateway, call management, and trunk

interconnection. Features can be controlled using licenses. Different service licenses are

selected to meet varying networking requirements. The following table lists license functions.

License Name Depends on Description

AR Voice

Value-added

Service License

None This license is mandatory for all voice functions. Other

licenses can be loaded only after this license is loaded.

Provide the function of direction access by end

users, including POTS and ISDN terminals.

Provide the SIP AG function that uses the SIP

protocol to process direct access from POTS and

ISDN terminals.

Support the TDM PBX VE1 access mode, and use

the SIP protocol to process access from VE1

phones.





64

License Name Depends on Description

CM&BEST

License

AR Voice

Value-added

Service

License

The call manager (CM) feature implements user

management and call management, for POTS phones,

ISDN phones, and SIP UEs. This feature supports all

new services of the PBX. Some new services, such as

IVR, require service licenses. Trunks, such as AT0, E1,

and SIP, are supported.

The BEST feature provides internal call and

inter-office call processing of the local branch when the

IP WAN is faulty. The BEST feature supports only

basic call functions, and does not support PBX

value-added services.

In a branch, the BEST feature can provide call

service for analog phones, IP phones or TDM

PBXs.

Inter-office calls can be routed through a SIP trunk

to the IP WAN or through an AT0 or E1 trunk to the

PSTN.

The E1/AT0 trunk takes effect only when the SIP

trunk is faulty. When the SIP trunk recovers,

inter-office calls are no longer routed through the

E1/AT0 trunk.

When the SIP trunk is working normally, calls in a

branch are processed by the headquarters CM after

being routed to the headquarters CM through the

SIP trunk.

CT (Call Trunk)

License

AR Voice

Value-added

Service

License

The call trunk (TG) feature provides interconnection

between trunks and converts packets of different

protocols. It does not provide direct access to terminals.

Support AT0, E1, and SIP trunk ports. These trunk

ports are peer-to-peer for the CT.

Support the conversion from H.323 to SIP and from

R2 to SIP and the codec conversion.

IVR (Interactive

Voice Response)

License

CM&BEST

License or CT

(Call Trunk)

License

The IVR automatic connection service is similar to an

attendant position. When a user dials the IVR service

access code, the IVR system plays a voice message,

promoting the user to dial an extension number or the

main number. The user can dial the extension number

to start a basic call, or dial the main number to trigger

the simultaneous ringing, sequential ringing, line

selection (selecting the next user in accordance with

certain rules) services. When multiple users call the

IVR service, subsequent users wait in the call queue

and hear the call queuing announcement.





65

11.1 AR Inter-Branch Voice Communication Solution

Huawei's voice communications system has three deployment models: centralized call control

model, distributed call control model, and mixed call control model.

Select the deployment model according to the enterprise scale, branch distribution, and

expansion plan based on the following factors:

1. Locations of the headquarters and branches. If the headquarters and branches are in the

same administrative region (they have the same PSTN area code), the centralized call

control model is recommended. In the model, all voice users in the enterprise

register with the headquarters, implement voice interconnection over the headquarters,

and initiate inter-office calls over the headquarters. If the headquarters and branches are

located in different PSTN areas, branches are connected to the headquarters through the

enterprise's intranet. The distributed call control model is recommended. In this model,

each branch processes internal calls and local inter-office calls.

2. Staff distribution and traffic volume between the headquarters and branches. If each

branch has a few employees and low voice traffic volume, it is recommended that the

centralized call control model be used to reduce workload on data configuration and

network maintenance in branches. If the number of employees in branches approximates

to that in the headquarters, it is recommended that the distributed call control model be

used to reduce bandwidth consumption between the headquarters and branches.

3. Voice service deployment. If value-added voice services are controlled by the

headquarters, the centralized call control model is recommended so that fewer devices

need to be deployed in branches. If branches need to control value-added voice services,

the distributed call control model is recommended.

4. Bandwidth and QoS guarantee. If links between the headquarters and branches can

provide sufficient bandwidth and QoS guarantee for voice services, the centralized call

control model is recommended. If links between the headquarters and branches cannot

provide sufficient bandwidth or QoS guarantee, the distributed call control model is

recommended because it reduces bandwidth consumption between the headquarters and

branches.

The call control models are described in the following sections.

11.1.1 Centralized Call Control Model

If the headquarters and branches of an enterprise are in the same area, the headquarters

connects to branches over a metropolitan area network (MAN). The CM and CT features are

deployed in the headquarters, and the AG feature is deployed in the branches. That is, the AR

router in the headquarters is used as an IP PBX, and the AR routers in the branches as SIP

AGs.

All voice users in the headquarters and branches register with the AR in the headquarters. The

AR in the headquarters provides call control services (CM) for all users, and connects

enterprise users to the local telecom operators (CT). Egress routers of medium-sized branches

provide the survivable remote site telephony (SRST) and power outage survival functions to

enhance communication reliability.





66

Figure 11-1 Centralized call control deployment

The AR on the headquarters is used as the VoIP CM in the enterprise. All internal short

numbers and users are registered and managed in the AR. In the headquarters deployed with

narrowband voice services, the existing TDM PBXs can still be used to provide call services

for analog users, and connect to the AR through E1/FXO ports. In this case, the AR router

needs to load the CT license to implement trunk interconnection.

The AR in a branch is used as a voice gateway to provide the AG function, and implement the

conversion from analog signaling to SIP signaling. For a branch with high reliability

requirement, the BEST feature can be loaded to provide basic call services for local

phones when the IP MAN is faulty.

The IP PBX in the headquarters is configured as follows:

interface Ethernet2/0/0 //Configure an IP address for an IP PBX upstream port.

ip address 192.168.1.1 255.255.255.0

#

voice

voip-address signalling interface Ethernet2/0/0 192.168.1.1 //Configure the

signaling address pool.

voip-address media interface Ethernet2/0/0 192.168.1.1 //Configure the media

address pool.

#

enterprise hw //Create the enterprise hw

dn-set local //Create the local DN set.

#

sipserver //Configure a SIP server.





67

signalling-address ip 192.168.1.1 port 5060 //Set the SIP server signaling address

to 192.168.1.1, and set the signaling port number to 5060

media-ip 192.168.1.1 //Set the SIP server media address to 192.168.1.1

register-uri huawei.com //Set the SIP server URI to huawei.com

home-domain huawei.com //Set the home domain URI of the SIP server to huawei.com

#

callprefix 2 //Create call prefix 2.

prefix 2

enterprise hw dn-set local //Bind the enterprise hw and DN set local to the call

prefix.

call-type category basic-service attribute 0 //Set the call attribute to local

call.

digit-length 4 4 //Set the maximum number length to 4, and the minimum number length

to 4.

#

callprefix 3

prefix 3

enterprise hw dn-set local

call-type category basic-service attribute 0

digit-length 4 4

#

pbxuser 2222 sipue enterprise hw //Configure a PBX user 2222, and set its user type

to SIPUE and enterprise to hw.

sipue 2222 //Configure the SIPUE ID.

telno country-code 86 area-code 25 2222 //Configure a phone number for the PBX

user.

dn-set local //Specify the DN set for the PBX user.

call-right in international-toll out international-toll //Configure the call-in

and call-out rights for the PBX user.

#

pbxuser 2223 sipue enterprise hw

sipue 2223

telno country-code 86 area-code 25 2223

dn-set local

call-right in international-toll out international-toll

#

pbxuser 3000 pots enterprise hw //Configure a PBX user 3000, and set its type to

SIPUE and enterprise to hw.

port 1/0/0 //Specify the physical port to which the PBX user is bound.


dn-set local


#

pbxuser 3001 pots enterprise hw

port 1/0/1


dn-set local


#

pbxuser 3002 sipue enterprise hw

sipue 3002


dn-set local


#





68

return

The SIP AG in a branch is configured as follows:

interface Ethernet2/0/0 //Set the IP address for a SIP AG upstream port.

ip address 192.168.1.2 255.255.255.0

#

voice

voip-address signalling interface Ethernet 2/0/0 192.168.1.2

voip-address media interface Ethernet 2/0/0 192.168.1.2

#

sipag 1 //Create the SIP AG interface 1.

signalling-addr 192.168.1.2 5060 //Set the signaling IP address of the SIP AG

interface to 192.168.1.2, and set the signaling port number to 5060.

media-addr 192.168.1.2 //Set the media IP address of the SIP AG interface to

192.168.1.2.

primary-proxy-addr static 192.168.1.1 5060 //Set the address of the active proxy

server to 192.168.1.1, and set the signaling port number to 5060.

home-domain huawei.com //Set the home domain name to huawei.com.

#

sipaguser 1 port 1/0/0 //Create a SIP AG user and set its interface number.

base-telno 2222 //Configure the telephone number of the SIP AG user.

agid 1 //Set the SIP AG interface associated with the SIP AG user to 1.

#

sipaguser 2 port 1/0/1

base-telno 2223

agid 1

#

return

11.1.2 Distributed Call Control Model

If the headquarters and branches of an enterprise are in different areas and the branches

connect to the headquarters over a private IP network, the distributed call control model can

be used for the enterprise. The CM and CT features are deployed on ARs in the headquarters,

and the CM feature is deployed on ARs in the branches.

Voices users in the headquarters are registered on the AR (CM) in the headquarters, and voice

users in each branch are registered on the AR (CM) in the branch. Voice traffic is transmitted

between the headquarters and branches through voice routes. The AR of the headquarters

provides voice routes (CT) so that users in different branches can call each other.





69

Figure 11-2 Distributed call control deployment

The IP PBX in the headquarters is configured as follows:

interface Serial2/0/0

link-protocol ppp

ip address 192.168.1.1 255.255.255.0

#

voice

voip-address signalling interface Serial 2/0/0 192.168.1.1

voip-address media interface Serial 2/0/0 192.168.1.1

pbx default-country-code 86

pbx default-area-code 25

#

enterprise hw

dn-set local

#

sipserver

signalling-address ip 192.168.1.1 port 5060

media-ip 192.168.1.1

register-uri huawei.com

home-domain huawei.com

#

trunk-group at0 fxo //Configure an AT0 trunk group.


call-right in international-toll //Configure the call-in right for the trunk group.





70

call-right out international-toll //Configure the call-out right for the trunk

group.

trunk-at0 1/0/4 default-called-telno 22223000 reversepole-detect disable //Add

a trunk to the trunk group.

#

trunk-group sipip sip no-register //Configure a SIP trunk group.


call-right in international-toll

call-right out international-toll

signalling-address ip 192.168.1.1 port 5070 //Set a signaling address and signaling

port number.

media-ip 192.168.1.1 //Set a media address.

home-domain huawei.com //Set the home domain name of the trunk group to huawei.com.

register-uri huawei.com //Set the register URL of the trunk group to huawei.com.

peer-address static 192.168.2.1 5070 //Set the remote IP address of the trunk

group to 192.168.2.1, and set the port number to 5070.

#

callprefix 9

prefix 9



digit-length 1 15

destination-location inter-office

callroute trunkgroup1 at0

#

callprefix 2222

prefix 2222



digit-length 8 8

#

callprefix 20000

prefix 20000



digit-length 5 20

destination-location inter-office

callroute trunkgroup1 sipip //Configure a call route.

#


port 1/0/0


dn-set local


service-right call-transfer enable

#


port 1/0/1


dn-set local


#

afterroute-change 9 //Configure post-routing number change.

callprefix 9

trunk-group at0 //Bind the AT0 trunk group to the call route.





71

calling party no-change //Set the calling number change rule: not to change the

calling number.

called del 7 1 //Configure the called number change rule: to delete the seventh

digit from the called number.

#

afterroute-change 20000

callprefix 20000

trunk-group sipip

calling party no-change

called del 7 5

#

Return

For IP PBX configurations in a branch, refer to the IP PBX configurations in the

headquarters.

11.1.3 Hybrid Call Control Model

An enterprise may have many branches. Some branches are in the same area and

communicate through an IP MAN; some branches are in different areas and communicate

through an IP leased line. The hybrid call control model is recommended for this enterprise. If

the enterprise has multiple branches in the same area, the enterprise can deploy the CM

feature in a large branch and deploy egress routers in other branches as AGs. Voice users in

this area are registered on the AR (CM) of the large branch. The egress routers of branches in

other areas load the CM feature. Voice users in these branches are registered on the ARs in

respective branches.

In the hybrid call control model, the communication mechanism used between branches in the

same area is the same as that used in the centralized call control model, and the

communication mechanism used between branches in different areas is the same as that used

in the distributed call control model.

11.2 AR Connecting to an IMS/NGN Network as AG

11.2.1 Market Positioning and Intended Customers

AR series routers are the mid-range-and-low-end enterprise routers that provide VoIP services.

Integrative VoIP module is an important feature of AR series routers. This solution aims at

enterprise users. Due to the performance limits, this solution is targeted at the resale VoIP

market.





72

11.2.2 Network Topology and Solution

Figure 11-3 Topology of AG deployment in an enterprise

IMS

Network

FAX Phone

Enterprise

NGN/IMS

Primary

SBC

Secondary

SBC

SIP

AR(AG)

As shown in Figure 11-3, the AR is used as a voice AG device. Register messages from voice

users are transmitted to the IMS. The AR connects voice users to the IMS, and forwards all

the process signaling and media streams to the IMS for processing and routing.

Upstream ports of the AR connect to the IMS or NGN network using the SIP.

The AR connects to POTS users using the FXS ports.

An AR can function as both an AG and egress router of an enterprise network to provide voice

and data services. This reduces costs for the enterprise. In carrier resale projects, voice service

is planned and managed by the carrier, and the enterprise does not need to operate and

maintain the network. Services on the AG are provided by the carrier's core network;

therefore, the carrier manages services in a uniform manner and the operation and

maintenance are simple.

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