Internet Telephony: VoIP, SIP & more

Shivkumar KalyanaramanRensselaer Polytechnic Institute

1

Internet Telephony: VoIP, SIP & more

: “shiv rpi”

Shivkumar Kalyanaraman

Adapted from slides of Henning Schulzrinne, Doug Moeller


2

Telephony: history and evolution IP Telephony: What, Why & Where?

Adding interactive multimedia to the web Being able to do telephony on IP with a variety of devices Consumer & business markets Key element of convergence in carrier infrastructure

Basic IP telephony model Protocols: SIP, H.323, RTP, Coding schemes, Megaco Future: Invisible IP telephony and control of appliances

Overview


3

What is VoIP? Why VoIP?

Where is VoIP Today?


4

What is VoIP?

VoIP = “Voice over IP” Transmission of telephony services via IP infrastructure => need history/concepts reg. both “telephony” (or “voice”) and “IP”

Complements or replaces other Voice-over-data architecture Voice-over-TDM Voice-over-Frame-Relay Voice-over-ATM

First proprietary IP Telephony implementations in 1994, VoIP-related standards available 1996 Buzzwords related to VoIP: H.323 v2, SIP, MEGACO/H.248, Sigtrans


5

What is VoIP? Protocol Soup

H.323

SIPMGCPSGCP

H.GCP

Megaco

IPD

C

H.245

SDP

MDCP

SigtransVPIM

Q.SIG

“The nice thing about standards is that you have so many to choose from; furthermore, if you do not like any of them, you can just wait for next year’s model.” [Tanenbaum]


6

Telephony over IP standards bodies ITU - International Telecommunication Union

http://www.itu.org IETF - Internet Engineering Task Force.

http://www.ietf.org ETSI - European Telecommunications Standards Institute

http://www.etsi.org/tiphon ANSI - American National Standards Institute

http://www.ansi.org TIA - Telecommunications Industry Association

http://www.tiaonline.org IEEE - Institute for Electrical and Electronics Engineers

http://www.ieee.org


7

Why VoIP? Telephony: Mature IndustryAT&T Divestiture


8

Why VoIP: Price/call plummeting due to overcapacity

AT&T Divestiture

1996 deregulation


9

Relevant Telecom Industry Trends 1984: AT&T breakup: baby bells vs long distance carriers 1996: Telecom deregulation, Internet takeoff Late 1990s: explosion of fiber capacity in long-distance + many new

carriers Long-distance prices plummet Despite internet, the last-mile capacity did not grow fast enough

2000s: shakeout & consolidation in developed countries Wireless substitution in last mile => cell phone instead of land-lines

Developing countries leap frog to cell phones 3G, WiMax => broadband, VoIP & mobility

Broadband rollouts happening slowly, but picking up steam now. Cable offering converged & bundled services:

digital cable, VoIP, video Recent mergers: AT&T (long-distance & data network provider) bought by

SBC (baby bell); Verizon/Qwest vs MCI saga…


10

Why VoIP ? Data vs Voice Traffic

Infrastructure convergence: Since we are building future networks for data, can we slowly junk the voice infrastructure and move over to IP?

Note: quantity quality value-added

Interactive svcs (phone, cell, sms)still dominate on a $$-per-Mbps basis


11

Trends: Total Phone vs Data Revenues


12

Motivations and drivers

Class 5switch

Class 4switch

Class 5switch

UsersUsers

PSTN

Packet networks

Data

Voice

H.323 gateway

ISDN Switch

Class-4/5 switches bulky, expensive. Incentive to switch to cheaper easily managed IP

Initial gateway between PSTN and Internet was H.323. Gateway did signaling, call control, translation in one box. Not scalable.


13

Voice Over IP Marketplace Drivers

Rate arbitrage declining but still has importance as cost driver TDM origination and termination with IP transport in the WAN International settlement and domestic access cost avoidance

Enterprises seeking to save on intra-company calls and faxes on converged network Emergence of native IP origination environments

IP PBX, IP Phones, Soft Phones, Multimedia on the LAN 3G Wireless, Broadband Networks

Companies: web-based call centers/web callback/e-commerce with IP Enablement New network-based IP features and services

Hosted IP PBX/IP Centrex , Unified Messaging, Multimedia Conferencing Presence: Mobility, Follow me, Teleworker, Voice Portal Services, WiFi

Technology maturing with open standards for easier, faster innovation Converging Local, long-distance (LD) and data services


14

VoIP Volumes Are Accelerating While Adoption of Applications is Growing

020000400006000080000

100000120000140000160000180000200000

2001 2002 2003 2004 2005 2006 2007

VoIP VPN Traffic

Source: Probe Research Inc.: Reaching the Big Guys + Global Enterprise Forecast. September 2002

0

50000

100000

150000

200000

250000

300000

350000

2001 2002 2003 2004 2005 2006 2007

Virtual PBX + Managed IP PBX traffic

M ofMinutes

M ofMinutes

North America Rest of the World

North America Rest of the World

Enterprise Adoption of VoIP / IPT Applications

• VoIP VPNs will continue to be driven by increased IPT deployments in larger enterprises, coupled with economic benefits accruing, especially for MNCs

• IPT Deployments are the leading edge market driver for the development of converged LANs and WANs

Source: Giga Group, "Next Generation IP Telephony Applications Deliver Strategic Business Value", October 20, 2003

Respondents


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Drivers Are Evolving From Cost Savings to Added Business Value…

Gartner Group, Sept. 16, 2003

Business Case

Justification Based on Business

Value

V3 Apps

V3 Apps - e.g. Unified Communications, Application Integration With Communications

V2 Apps

Business Case Justification

Based on Investment Protection

V2 Apps - e.g. Call Center Functions, Messaging, Administration Tools and Reports

Percentage IP Phones Performing Functions Other Than

POTS

2003 2007200620052004 20082002

Business Case

Justification Based on

Cost Savings

V1 Apps

V1 Apps - e.g. IP-PBX, Basic Call Functions, Branch offices, Toll-bypass

Cost Savings• Toll By-Pass• Effective Use of Bandwidth • Personnel / Staffing Efficiencies• Less Expensive Moves, Adds Changes• Convergence / Consolidation• Decreased Capital• Upgrading to an IP PBX

Increased Investment Protection• Contact Center Functions• Future Proofing Infrastructure• Leveraging embedded infrastructure with a

phased roll-out• Networking Expertise for Integration From

Concept to Deployment

Optimized Business Value• Services over IP• Consistent Client / User Experience • Integrated Infrastructure• End-to-End Interoperability


16

Summary: Why VoIP? Cost reduction:

Toll by-pass WAN Cost Reduction Lowered Infrastructure Costs

Operational Improvement: Simplification of Routing Administration LAN/Campus Integration Policy and Directory Consolidation

Business Tool Integration: Voice mail, email and fax mail integration Mobility enabled by IP networking Web + Overseas Call Centers Collaborative applications New Integrated Applications

3Cs: “Convergence” & “Costs” & “Competition”


17

Where is VoIP? Consumer VoIP Markets

Convergence & Competition Vonage: pure VoIP CLEC (300K subscribers) Cable companies:

Eg: Time Warner (220K subscribers and signing on 10K per week (end of 2004)):

Bundled with digital cable services Skype (computer-computer p2p VoIP): tens of millions…

Also has a WiFi service & a product co-developed by Motorola (over 3G networks)

Long-distance providers: AT&T CallVantage Local (ILECs): Verizon

Future: convergence of VoIP + WiMax (802.16) as a open low-cost competitor to 3G wireless (closed system) Combines: broadband Internet, mobility and VoIP


18

Consumer VoIP over broadband

Broadband Infrastructure

ResidentialMedia Gateway

Traditional phone

Media Gateway Controller

Signaling and media gatewaysTo reach PSTN or other networks


19

Consumer VoIP at home with cable

PacketCable standard with DOCSIS 1.1 access infrastructure

MTA(Media Terminal Adapter)

Cable Modem

Call Management Server

MGCCable Modem Term. Sys.

MediaGateway

SignalingGateway

Other access mechanisms will similarly hand over to an MGC


20

Consumer VoIP: AT&T CallVantage

New consumer services: Personal conferencing: earlier available to businesses only

Prepaid Calling cards offering personal conferencing Portable TA (terminal adaptor): can plug into any ethernet

jack or WiFi (eg: many hotels providing free internet) Universal messaging: voice messages in email LocateMe, Do-Not-Disturb, Unified Portal


21

Skype: p2p VoIP over Internet

Skype is entirely peer-to-peer and is equivalent to two H.323 terminals or 2 SIP terminals talking to each other Provides a namespace Efficient coding of

voice packets Instant messaging with

voice Uses Kazaa-like p2p

directory + secure authentication (login server) and e2e encryption


22

VoIP over Wireless Cellular networks with 2.5G and 3G have packet services

1xRTT on 2.5 G EV-DO on 3G

The voice on these networks is circuit switched voice…

However, … Combined with bluetooth or USB interfaces, a PC-based VoIP software

can do VoIP anywhere there is cellular coverage. Or Cellphone can be a SIP terminal

Near Future: VoIP over WiMax (802.16) and WiFi (802.11) networks


23

Enterprise: Private Branch Exchange (PBX)

7043

7040

7041

7042

External line

Telephoneswitch

Private BranchExchange

212-8538080

Anotherswitch

Corporate/Campus

InternetCorporate/Campus LAN

Post-divestiture phenomenon...


24

Enterprise VoIP: Yesterday’s networksCircuit Switched Networks (Voice)

Packet Switched Networks (IP)

PBXPBX

COCO

CO

Router

RouterRouter

Router

Router

Headquarters Branch Offices


25

Enterprise VoIP: Today’s networksToll by-pass

Circuit Switched Networks (Voice)

Packet Switched Networks (IP)

PBXPBX

COCO

CO

Router

RouterRouterRouter

Router



26

Enterprise VoIP: Tomorrow’s networksUnified/Converged Networks

Unified Networks (Voice over IP)

Router

RouterRouter

Router

Router

COCO

Legacy PSTN



27

• VoIP infrastructure is converged onto a single IP/ MPLS network

• Open standards architecture based on SIP protocol

• Call Control Element manages all SIP signaling within our core network

• Access Agnostic: TDM, ATM, Frame, MIS, IP Enabled Frame and EVPN

• Border Elements: “translate” the multiple protocols into SIP, provide compression and security

• Provides secure, integrated voice / data / video access

• Flexibility to support future applications

• VoIP infrastructure is converged onto a single IP/ MPLS network

• Open standards architecture based on SIP protocol

• Call Control Element manages all SIP signaling within our core network

• Access Agnostic: TDM, ATM, Frame, MIS, IP Enabled Frame and EVPN

• Border Elements: “translate” the multiple protocols into SIP, provide compression and security

• Provides secure, integrated voice / data / video access

• Flexibility to support future applications

AT&T’s Integrated Infrastructure Supports Multiple Endpoints, Access Technologies and Application Services

Common VoIP Connectivity Layer

H.323 endpoints PSTN

SIP endpoints

NG Border Elements

SIP Border Elements

H.323 Border

Elements

MGCP Border

Elements

MGCP endpoints

IP/MPLS Converged Network

AT&T Call Control Element

Voice Applications: IP Centrex, IP Call Center and Distant Worker


28

Outbound Call• IP to Circuit Switched

NetworkAdjunct

Softswitch

App.Server Gateway

CustomerRecords

MediaServer

App.Server

Inbound Call• Circuit Switched to IP

800 Call• Circuit Switched to IP

Redirect Call• Circuit Switched to IP

SDN Call• IP to Circuit Switched

VoIP Network UtilitiesEnsure Seamless Operations

Circuit Switched Network

IP Network


29

IP-enabled circuit switches

PBX with VoIP trunk card trunk between PBX

Key system or PBX with VoIP line card for IP phones

VoIP Gateway

CO

Switch


30

Telephony-enabled packet networks

RouterVoIP

Gateway

Enterprise Router with telco interfaces T1/PRI BRI

Branch office router with telco interfaces BRI Analog trunk/line

Analog “dongle” a few analog lines

for fax/phone

Central Office


31

VoFR (Voice over Frame Relay)

FRF.11 standard Allows for G.711, 729, 728, 726, and 723.1 Signaling is done by transporting CAS natively or

CCS as data Has support for T.30 Fax, and Dialed Digits natively

PBX

Switch

SwitchSwitch

PBXVFRAD

VFRAD

Router


32

Voice over Packet: Market Forecast – North America


33

Telephony: History, Review & Trends


34

VoIP: Where Does it Fit in Trends ? Phase 1: Analog Networks:

Voice carried as analog signal Phase 2: Digital Networks & the rise of the Internet

Network is digital: analog conversion at end systems Benefits: [Noise , capacity] Egs: TDM and T-hierarchy (T1, T3, SONET etc)

Used as the base for the internet & private data networks Phase 3: Voice-over-X:

Voice over Packets: VoFR, VoIP Key: Voice moves to a higher layer (from layer 1) I.e. an app over a frame relay, ATM or IP network

VoIP Sales pitch: Convergence, Choice, Services, Integration with Web applications

[Better chance of convergence compared to earlier attempts: ISDN, B-ISDN]


35

Public Telephony (PSTN) History 1876 invention of telephone 1915 first transcontinental telephone (NY–SF) 1920’s first automatic switches 1956 TAT-1 transatlantic cable (35 lines) 1962 digital transmission (T1) 1965 1ESS analog switch 1974 Internet packet voice 1977 4ESS digital switch 1980s Signaling System #7 (out-of-band) 1990s Advanced Intelligent Network (AIN)


36

PSTN Evolution

Full Mesh Office Switched

W/ HierarchyOffice Switched


37

AT&T Telephony Hierarchy

1

109 8

7

3

2

45

6

1 2 3

1 2 3

1 2 3

65 66 67

228 229 230

1298 1299 1300

1 2 3 4 519,000

200 million telephones

19,000 endoffices

1300 tolloffices

230 primaryoffices

67 sectionaloffices

10 regionaloffices(full mesh)

Source: Computer Networks, Andrew S. Tanenbaum

Class 5

Class 4

Class 3

Class 2

Class 1


38

PSTN early days 40s-60s

User AUser B

Local Office

Tandem Office

Local Office

1. In-band signaling: voice and control channel same

2. Complex and dedicated hardware

3. Hard to add new apps like caller-id, 800 calling etc


39

Advanced Intelligent Network

User AUser B

Voice Network

Local Office

Signaling Network

Customer Info forAdvanced services

•Out-of-band signaling•Introduce adv services like caller-id easily•Reduced wastage of circuits in voice network•Signaling could be over a packet network •E.g. SS7 stack

Sometimes also called Intelligent Network, arrival of services other than voice


40

The PSTN – Architecture PSTN – Public Switched Telephone Network Uses digital trunks between Central Office switches (CO) Uses analog line from phones to CO

Digital Trunks

Analog line

CentralOffice (CO)

Analog Digital Analog


41

The PSTN – Digitization

Voice frequency is 100 - 5000 Hz, with the main portion from 300 – 3400 Hz

Nyquist Theorem states that sampling must be done at twice the highest frequency to recreate. 4000 Hz was chosen as the maximum frequency, thus sampling at 8000 Hz

PCM = 8kHz * 8 bits per sample = 64 kbit/s


42

Quantization


43

Companding


44

The PSTN – Digitization

The PCM encoding used in the PSTN is standardized as G.711 by the ITU

Each sample is represented by one byte The voice signal is companded to improve voice

quality at low amplitude levels (Which most conversation is at)

The ITU standards for companding are called A-law and u-law

G.711 A-law is used in Europe G.711 -law is used in the US and Japan


45

The PSTN – Digital Voice Transmission The digital trunks between the COs are based upon the T-

carrier system, developed in the 1960s Each frame carries one sample (8 bits) for each 24 channels,

plus one framing bit = 193 bits 193 * 8000 (samples/sec) = 1.544 Mbit/sec = T-1

Channel 1

Channel 2

Channel 3

Channel 24

…

Channel1

Channel2

Channel3

Channel24…

Framing Bit

1 D4 Frame

TDM


46

The PSTN – Architecture, Switches PSTN – Public Switched Telephone Network As the name says, it’s switched… Each conversation requires a channel switched throughout the network Circuit setup uses a separate out-of-band intelligent network (SS7)

1. Call is requested 3. Channel is established 2. Call is accepted


47

Legacy Digital Circuit Switch

• Centralized Intelligence

• Proprietary Code

• Proprietary service deployment

• Very expensive

TrunkCard

TrunkCard

TrunkCard

LineCard

LineCard

LineCard

Switch Controller

Next Switch

Next Switch

Next Switch

SS7 Network


48

What’s the difference between a Class 5 and a Class 4 switch?

Class 5 Located at the edge of the

network Trunk to Line/Line to Line Aprox. 30,000 deployed Services: Caller ID, call

waiting, voice mail, E911, billing, etc.

Ex: Lucent 5ESS, Nortel DMS, Siemens EWSD

Class 4 Located in the Core of the

network Trunk to Trunk Aprox. 800 deployed Services: call routing,

screening, 800 services, calling cards, etc.

Ex: Lucent 4ESS, Nortel DMS, Siemens


49

PSTN

The PSTN – NANP

NANP – North American Numbering Plan 3 digits area code + 3 digits office code + 4 digits phone Each Local Exchange Carrier (LEC) switch are assigned a

block of at least 10,000 numbers The Inter-Exchange Carrier (IXC) switches are responsible for

transmitting long distance

IXC212

LEC555

4210

(212) 555 4210


50

The PSTN – Call Routing Both NANP and International Numbering Plan – E.164, use

prefix-based dialing

408 5644555212PSTN

1+212+555+5644

The first LEC receives a call, seeing ‘1’ as the first digit and then passing the call on to theIXC switch. The IXC then routes the call to the remote IXC responsible for ‘212’

555+5644

The ‘212’ IXC looks at the office code and passes it on to the ‘555’ LEC switch

5644

The ‘555’ LEC switch then checks the station code and signals the appropriate phone

SS7


51

Telephone System Summary

Analog narrowband circuits: home-> central office 64 kb/s continuous transmission, with compression

across oceans-law: 12-bit linear range -> 8-bit bytes

Everything clocked a multiple of 125 s Clock synchronization framing errors

AT&T: 136 “toll”switches in U.S. Interconnected by T1, T3 lines & SONET rings

Call establishment “out-of-band” using packet-switched signaling system (SS7)


52

Telecommunications Regulation History FCC regulations cover telephony, cable, broadcast TV, wireless

etc

“Common Carrier”: provider offers conduit for a fee and does not control the content Customer controls content/destination of transmission & assumes

criminal/civil responsibility for content

Local monopolies formed by AT&T’s acquisition of independent telephone companies in early 20th century Regulation forced because they were deemed natural monopolies (only one

player possible in market due to enormous sunk cost) FCC regulates interstate calls and state commissions regulate intra-state and

local calls Bells + 1000 independents interconnected & expanded


53

Deregulation of telephony 1960s-70s: gradual de-regulation of AT&T due to

technological advances Terminal equipment could be owned by customers (CPE)

=> explosion in PBXs, fax machines, handsets Modified final judgement (MFJ): breakup of AT&T into

ILECs (incumbent local exchange carrier) and IXC (inter-exchange carrier) part

Long-distance opened to competition, only the local part regulated…

Equal access for IXCs to the ILEC network 1+ long-distance number introduced then…

800-number portability: switching IXCs => retain 800 number

1995: removed price controls on AT&T


54

US Telephone Network Structure (after 1984)

Eg: AT&T, Sprint, MCI

Eg: SBC, Verizon, BellSouth


55

Telecom Act of 1996 Required ILECs to open their markets through unbundling of

network elements (UNE-P), facilities ownership of CLECs…. Today UNE-P is one of the most profitable for AT&T and other long-distance

players in the local market: due to apparently below-cost regulated prices…

ILECs could compete in long-distance after demonstrating opening of markets Only now some ILECs are aggressively entering long distance markets CLECs failed due to a variety of reasons… But long-distance prices have dropped precipitously (AT&T’s customer unit

revenue in 2002 was $11.3 B compared to 1999 rev of $23B) ILECs still retain over 90% of local market Wireless substitution has caused ILECs to develop wireless business units VoIP driven cable telephony + wireless telephony => more demand elasticity for

local services


56

VoIP Technologies


57

IP Telephony Protocols: SIP, RTP

Session Initiation Protocol - SIP Contact “office.com” asking for “bob” Locate Bob’s current phone and ring Bob picks up the ringing phone

Real time Transport Protocol - RTP Send and receive audio packets


58

Inside the Endpoint: Data-plane … I.e.after signaling is done… Consists of three components:

UserUser

A/DCodec

A/DCodec

IPGateway

IPGateway

User speaks into microphone, either PC attached, regular analogue phone or IP phone

Device digitizes voice according to certain codecs:

G.711 / G.723.1 / G.729 ...

Voice gets transmitted via RTP over an IP infrastructure


59

Internet Multimedia Protocol Stack


60

PreambleDestination

AddressData Pad Checksum

SourceAddress

Inter-framegap

Start of framedelimiter

Length orEthertype

12 7 1 6 6 2 0-1500 0-46 4

Ethernet Frame

DestinationAddress

SourceAddress

HeaderChecksum

DataFlags &

Frag OffsetTotal

LengthPacket

IDOptions(if any)

1 1 2 2 2 1 1 2 4 4 0-40 0-1480

IP packet

Version &header length

TOS TTL

Protocol

SourcePort Number

DestinationPort Number

UDP length UDP checksum

2 2 2 2 0-1472

UDP datagram

Version,flags & CC

SequenceNumber

Timestamp

1 1 2 4 4 0-60 0-1460

RTP datagramSynchronization

Source IDPayload

TypeCSRC ID(if any)

Codec Data

Data

Packet Encapsulation


61

RTP – Real-time Transport Protocol

Byte 1: Version number, padding yes/no, extension y/n, CSRC count

Byte 2: Marker, Payload type Bytes 3,4: Sequence number for misordered and lost packet

detection Bytes 5-8: Timestamp of first data octet for jitter calculation Bytes 9-12: Random syncronization source ID Bytes 13-x: Contributing Source ID for payload Codec Data: the actual Voice or Video bytes

Version,flags & CC

SequenceNumber

Timestamp

1 1 2 4 4 0-60 0-1460


Source IDPayload

TypeCSRC ID(if any)

Codec Data


62

RTCP – Real-time Transport Control Protocol

RTCP is sent between RTP endpoints periodically to provide: Feedback on quality of the call by sending jitter,

timestamps, and delay info back to sender Carry a persistent transport-level identifier called the

canonical name (CNAME) to keep track of participants and synchronize audio with video

Carry minimal session information (like participant IDs), although signaling protocols do this much better

RTCP is mandatory for multicast sessions and for many point-to-point protocols, but some boxes don’t implement it

Uses another UDP port (usually RTP’s port + 1)


63

SIP


64

Signaling: VoIP Camps

ISDN LAN conferencin

g

IP

H.323

I-multimediaWWW

IP

SIP

Call AgentSIP & H.323

IP

“Softswitch” BISDN, AIN

H.xxx, SIP

“any packet”

BICC

Conferencing Industry

Netheads“IP over

Everything”

Circuit switch

engineers “We over

IP”

“Convergence” ITU

standards

Our focus


65

H.323 vs SIP H.323: ITU standard

Derived from telephony protocol (Q.931) Follows ISDN model: same control message sequences Interfaces well with telephony services (H.450, Q.SIG)

SIP: IETF standard Derived from HTTP style signaling, Simple and interfaces well with IP networks, instant

messaging (IM) Services are not explicitly exposed to protocol Well-defined methods can be used to design services: most

telephony services have analogs in the SIP world today SIP is gathering market share rapidly


66

SIP

SIP

Audio Codec

G.711

G.723

G.729

Video Codec

H.261

H.263

RTP RTCP

LAN Interface

TCP

IP

UDP


67

SIP functionality IETF-standardized peer-to-peer signaling protocol (RFC

2543): Locate user given email-style address Setup session (call) (Re)-negotiate call parameters Manual and automatic forwarding Personal mobility: different terminal, same identifier Call center: reach first (load distribution) or reach all

(department conference) Terminate and transfer calls


68

SIP Addresses Food Chain


69

Why is SIP interesting? SIP is IETF’s equivalent for H.323 to provide a peer-based signaling

protocol for session setup, management and teardown

Simple, did not inherit the complexity of ISDN Analogy: CISC architecture Though all services arent defined as in H.323, you can compose them

with primitives

Was designed with multimedia in mind Just requires a MIME type Tremendous flexibility – can add video, text etc to a voice session,

similar to what HTTP did to Internet content

Like H.323, can use SIP end-to-end with no network infrastructure (MGC etc.) – peer-to-peer

Lightweight can be embedded in small devices like handhelds


70

Are true Internet hosts

• Choice of application

• Choice of server

• IP appliances

Implementations

• 3Com (3)

• Columbia University

• MIC WorldCom (1)

• Mediatrix (1)

• Nortel (4)

• Siemens (5)

4

IP SIP Phones and Adaptors

1

3

Analog phone adaptor

Palmcontrol

2

54


71

SIP: Personal Mobility

Users maintain a single externally visible identifier regardless of their network location


72

Expand existing PBXs w/ IP phones

Transparently …


73

SIP as Event Notification Protocol


74

SIP: Presence


75

Light-weight signaling: Session InitiationProtocol (SIP)

IETF MMUSIC working group Light-weight generic signaling protocol Part of IETF conference control architecture:

SAP for “Internet TV Guide” announcements RTSP for media-on-demand SDP for describing media others: malloc, multicast, conference bus, . . .

Post-dial delay: 1.5 round-trip time (with UDP) Network-protocol independent: UDP or TCP (or

AAL5 or X.25)


76

SIP components

UAC: user-agent client (caller application) UAS: user-agent server: accept, redirect, refuse call redirect server: redirect requests proxy server: server + client registrar: track user locations user agent = UAC + UAS often combine registrar + (proxy or redirect server)


77

SIP-based Architecture

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

SIPH.323convertor

NetMeetingsip323

H.323

rtspd

SIP/RTSPUnified

messaging

RTSP media server

sipum

Quicktime

RTSP clients

RTSP

SIP conference

server

sipconf

T1/E1 RTP/SIP

Telephone

Cisco 2600 gateway

Telephoneswitch Web based

configuration

Web server


78

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Web based configuration

Web server

Call Bob

Example Call

• Bob signs up for the service from the web as “[email protected]”

• He registers from multiple phones

• Alice tries to reach Bob INVITE ip:[email protected]

• sipd canonicalizes the destination to sip:[email protected]

• sipd rings both e*phone and sipc

• Bob accepts the call from sipc and starts talking

ecse.rpi.edu


79

SIP Sessions “Session”: exchange of data between an association of

participants Users may move between endpoints Users may be addressable by multiple names Users may communicate in several different media SIP: enables internet endpoints to

Discover each other Characterize the session

Location infrastructure: proxy servers, invite/register… Name mapping and redirection services

Add/remove participants from session Add/remove media from session


80

SIP Capabilities

User location: determination of the end system to be used for communication;

User availability: determination of the willingness of the called party to engage in communications;

User capabilities: determination of the media and media parameters to be used;

Session setup: "ringing", establishment of session parameters at both called and calling party;

Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.


81

What SIP is not…

SIP is not a vertically integrated communications system. It is a component in a multimedia architecture.

SIP does not provide services. Rather, SIP provides primitives that can be used to

implement different services. For example, SIP can locate a user and deliver an opaque

object to his current location. SIP does not offer conference control services

… such as floor control or voting SIP does not prescribe how a conference is to be managed.


82

SIP Structure 3 “layers”, loosely coupled, fairly independent processing stages Lowest layer: syntax, encoding (augmented BNF) Second layer: transport layer.

Defines how a client sends requests and receives responses and how a server receives requests and sends responses over the network.

Third layer: transaction layer. A transaction is a request sent by a client transaction (using the

transport layer) to a server transaction … …along with all responses to that request sent from the server

transaction back to the client. The transaction layer handles application-layer retransmissions,

matching of responses to requests, and application-layer timeouts

The layer above the transaction layer is called the transaction user (TU).


83

SIP Design Choices


84

Proxy server

parliament.uk

[email protected]

Location Server

ge

org

e.w

.bu

sh d

ch

en

ey

@w

h

3. SIP/2.0 200 ok From: sip:dcheney@wh

4. SIP/2.0 100 OK From: sip:[email protected]

1. INVITE sip:[email protected] SIP/2.0 From: sip:[email protected]

5. ACK sip:[email protected] SIP/2.0 From: sip:[email protected]

6. ACK sip:dcheney@wh SIP/2.0 From: sip:[email protected]

2. INVITE sip:dcheney@wh SIP/2.0 From: sip:[email protected]

1 & 5

4

2 & 6

3

Proxy Server

us.gov

dcheney@wh


85

parliament.uk

[email protected]

us.gov

Redirect Server

Location Server

ge

org

e.w

.bu

sh d

ch

en

ey

@w

h

2. SIP/2.0 320 Moved temporarily Contact: sip:[email protected]

3. ACK sip:[email protected] From: sip:[email protected]

1. INVITE sip:[email protected] From: sip:[email protected]

6. ACK sip:[email protected] From: sip:[email protected]

4. INVITE sip:[email protected] From: [email protected]

5. SIP/2.0 200 OK To: [email protected]

1 & 3

2

5

4 & 6

Redirect Server

[email protected]


86

SIP Call Signaling

Media (UDP)RTP StreamRTCP Stream

SIPEndpoint

SIP + SDP (TCP or UDP)

Invite

RTP Stream

SIPGateway

Assumes Endpoints(Clients) know each other’s IP addresses

Signaling Plane

BearerPlane

200 OK

Ack

180 Ringing


87

PSTN to IP Call

PBXPSTN

External T1/CAS

Regular phone(internal)

Call 93971341

SIP server

sipd

Ethernet

3

SQLdatabase

4 7134 => bob

sipc

5

Bob’s phone

GatewayInternal T1/CAS(Ext:7130-7139)

Call 71342


88

IP to PSTN Call

Gateway(10.0.2.3)

3

SQLdatabase

2Use sip:[email protected]

Ethernet

SIP server

sipdsipc

1Bob calls 5551212

PSTN

External T1/CAS

Call 55512125

5551212

PBX

Internal T1/CASCall 85551212 4

Regular phone(internal, 7054)


89

Traditional voice mail system

Alice939-7063

Bob853-8119

Dial 853-8119

Phone is ringing

.. The person is not available nowplease leave a message ...

... Your voice message ...

Disconnect

Bob can listen to his voice mails by dialing some number.


90

SIP-based Voicemail Architecture

INVITE [email protected]

Alice

phone1.office.com

Bob

Alice calls [email protected] through SIP proxy.SIP proxy forks the request to Bob’s phone as well as to a voicemail server.

vm.office.com

The voice mail server registers with the SIP proxy, sipd



REGISTER [email protected]


91

Voicemail Architecture

v-mail

rtspd

Alice

vm.office.com;

200 OK

200 OK

CANCEL

SETUP

RTP/RTCP

phone1.office.com;

Bob

After 10 seconds vm contacts the RTSP server for recording.

vm accepts the call.Sipd cancels the other branch and ......accepts the call from Alice.Now user message gets recorded


92

IETF SIP Architecture Tour: RoundupRegistrar & Proxy or Redirect Server

Registrar & Proxy or Redirect Server

*Gateway*Gateway

*User Agent*User Agent*User Agent*User Agent*User Agent*User Agent

Media streams: RTP/RTCP (G.911, G.723.1, … )

PSTN,ISDN,ATM,etc

*Endpoints


93



*Gateway*Gateway



PSTN,ISDN,ATM,etc

*Endpoints

Interface to non-IP or H.323

networks

Interface to non-IP or H.323

networks

End-user devicesand network proxies


Conferencing does not need another

box (MCU)

Conferencing does not need another

box (MCU)

System Management• admission control• address

translation/forwarding• Firewall bypassing

System Management• admission control• address

translation/forwarding• Firewall bypassing


94



*Gateway*Gateway



PSTN,ISDN,ATM,etc

*Endpoints

Components of the SIP protocol suite:•SIP = almost all signaling, optional services, etc.•SDP = negotiation/capabilities •DNS = address translation•RSVP = QoS bandwidth guarantee

Components of the SIP protocol suite:•SIP = almost all signaling, optional services, etc.•SDP = negotiation/capabilities •DNS = address translation•RSVP = QoS bandwidth guarantee


95

SDP: Session Description Protocol Not really a protocol – describes data carried by other

protocols Used by SAP, SIP, RTSP, H.332, PINT. Eg:

v=0o=g.bell 877283459 877283519 IN IP4 132.151.1.19s=Come here, Watson!u=http://[email protected]=IN IP4 132.151.1.19b=CT:64t=3086272736 0k=clear:manhole coverm=audio 3456 RTP/AVP 96a=rtpmap:96 VDVI/8000/1m=video 3458 RTP/AVP 31m=application 32416 udp wb


96

Upcoming SIP Extensions (probable)

Call Admission Control Caller Preferences and Callee Capabilities Call Transfer SIP to ISUP mapping SIP to H.323 mapping Resource Management (QoS preconditions) Caller/Callee Name Privacy SIP Security Supported Options Header Session Timer Refresh Distributed Call State 3rd Party Call Control Early media for PSTN interoperability There are currently 47 drafts in the pipeline! 174 Drafts have expired


97

SIP Dialogs (RFC 3261)

A dialog represents a peer-to-peer SIP relationship between two user agents that persists for some time.

The dialog facilitates sequencing of messages between the user agents and proper routing of requests between both of them.

The dialog represents a context in which to interpret SIP messages.

A dialog is identified at each UA with a dialog ID, which consists of a Call-ID value, a local tag and a remote tag.

A dialog contains certain pieces of state needed for further message transmissions within the dialog.

Note: dialog is within SIP whereas sessions are outside SIP


98

UPDATE method (RFC 3311) INVITE method: initiation and modification of sessions.

INVITE affects two pieces of state: session (the media streams SIP sets up) and dialog (the state that SIP itself defines).

Issue: need to modify session aspects before the initial INVITE has been answered. A re-INVITE cannot be used for this purpose: impacts the state of the

dialog, in addition to the session. Ans: The UPDATE method

Operation: (Offer/Answer model) The caller begins with an INVITE transaction, which proceeds normally. Once a dialog is established, either early or confirmed, … … the caller can generate an UPDATE method that contains an SDP offer

for the purposes of updating the session. The response to the UPDATE method contains the answer. Similarly, once a dialog is established, the callee can send an UPDATE

offer


99

Locating SIP Servers (RFC 3263)

UA Proxy Remote Proxy UA I.e Go via proxies (per-domain) Issue: need to locate remote proxy (use DNS) DNS NAPTR (type of server) and SRV (server

URL) queries are used to locate the specific servers.

Different transport protocols can be used (TLS+TCP, TCP, UDP, SCTP)


100

SIP for instant messaging: IM (RFC 3428)

IM: transfer of (short) messages in near real-time, for conversational mode. Current IM: proprietary, server-based and linked to buddy

lists etc MESSAGE method: inherits SIP’s request routing and security

features Message content as MIME body parts Sent in the context of some SIP dialog (note: slightly different from pager mode: asynchronous) Sent over TCP (or congestion controlled transports): lots of

messaging volumes… Allows IM applications to potentially interoperate and also

provides SIP-based integration with other multimedia streams.


101

SIP compression (RFC 3486)

Cannot use DNS SRV and NAPTR techniques: non-scalable (only useful for specifying transport protocol options)

Use an application-level exchange to specify compression of signaling info sip:[email protected];comp=sigcomp Via: SIP/2.0/UDP

server1.foo.com:5060;branch=z9hG4bK87a7;comp=sigcomp SIGCOMP is the compression protocol


102

Device Configuration


103

SIP Scaling Issues


104

SIP Scaling (contd)

SIP Load Characteristics:


105

H.323


106

SIP vs H.323 vs Megaco


107

H.323 vs SIP

IP and lower layers

TCP UDPTPKT

Q.931 H.245 RAS RTCPRTP

Codecs

Terminal Control/Devices

Transport Layer

SIP SDPRTP

CodecsRTCP

Terminal Control/Devices

Typical UserAgent Protocol stack for Internet


108

SIP versus H.323

• Complex, monolithic design• Difficult to extend & update• Based on H.320 conferencing and

ISDN Q.931 legacy (“Bell headed”) • Powerful for video-conferencing

• Modular, simplistic design• Easily extended & updated• Based on Web principals (“Internet-

friendly”)• Readily extensible beyond telephony

Properties

• H.450.x series provides minimal feature set only, and not implemented by many

• Options and versions cause interop problems

• Slow moving

• Few real end-device features standard, and not implemented by many

• Many options for advanced telephony features

• Good velocity

Stds Status(end device)

•ITU-T SG-16 •IETF SIPStds Body

• Established now, primarily system level• Few H.323-based telephones• End-user primarily driven by Microsoft

(NetMeeting), Siemens, Intel

• Rapidly growing industry momentum, at system and device level

• Growing interest in SIP-phones and soft clients

IndustryAcceptance

H.323 SIP

H.323 and SIP are direct competitors in peer-level call control space


109

SIP-H.323: Interworking ProblemsEg: Call setup translation

Q.931 SETUP

Q.931 CONNECT

INVITE

200 OK

ACK

Terminal Capabilities

Terminal Capabilities

Open Logical Channel

Open Logical Channel

H.323 SIP

Destination address ([email protected])

Media capabilities (audio/video)

Media transport address (RTP/RTCP receive)

• H.323: Multi-stage dialing


110

H.323 Standard Series

System Control

H.245 Control

H.225 Call Setup

RAS Gatekeeper

Audio Codec

G.711

G.723

G.729

Video Codec

H.261

H.263

RTP RTCP

LAN Interface

TCP

IP

UDP

Data Interface

T.120


111

Internet Telephony Protocols: H.323


112

H.323 (contd) Terminals, Gateways, Gatekeepers, and Multipoint

Control Units (MCUs)


113

H.323 Model - Gatekeeper Routed Call

Call S

etup

/Sign

aling

Voice Channel

Gatekeeper

Endpoint

Gateway

Call Setup/SignalingCall

Con

trol

RAS Call Control

RAS


114

H.323 Model - Gatekeeper Direct Call

Call Setup/Signaling

Call Control

Voice Channel

RAS RAS

Gatekeeper

EndpointGateway


115

H.323 Call Signaling

Media (UDP)RTP StreamRTCP Stream

H.323Endpoint

H.245 (TCP)Open Logical Channel

H.225 (TCP)(Q.931)

Setup

Connect

Open Logical Channel & Acknowledge

RTP Stream

H.323Gateway

H.323v1 (5/96) - 7 or 8 Round TripsH.323v2 Fast Start (2/98) - 2 Round Trips

Assumes Endpoints(Clients) know each other’s IP addresses

Signaling Plane

BearerPlane

Alerting

Terminal Capability Set

Terminal Capability Set & Acknowledge

Terminal Capability Set Acknowledge

Open Logical Channel Acknowledge


116

ITU-T H.323 Architecture TourGate Keeper

(GK)

Gate Keeper(GK)

*Gateway (GW)*Gateway (GW)

*Terminal*Terminal

*Multipoint ControlUnit (MCU)


MultipointController

(MC)

MultipointProcessor

(MP)*Terminal*Terminal*Terminal*Terminal


PSTN,ISDN,ATM,etc

*Endpoints


117


(GK)

Gate Keeper(GK)


*Terminal*Terminal




(MC)

MultipointProcessor



PSTN,ISDN,ATM,etc

*Endpoints

Interface to non-IP networks

Interface to non-IP networks



ConferencingConferencing

System Management• zone management• b/w management &

admission control• address translation• centralized control

(“gatekeeper control mode”)

System Management• zone management• b/w management &

admission control• address translation• centralized control

(“gatekeeper control mode”)


118


(GK)

Gate Keeper(GK)


*Terminal*Terminal




(MC)

MultipointProcessor



PSTN,ISDN,ATM,etc

*Endpoints

Components of the H.323 protocol suite:•Q.931 = ISDN call signalling•H.225.0 = RAS (registration/admissions/status) gatekeeping functions

+ Call signalling channel (CS), contains Q.931•H.245 = Control channel (CC), negotiation/capabilities, logical signalling,

maintenance •H.450.x = Supplementary services (SS), transfer, hold, park, msg wait, … incomplete!

Components of the H.323 protocol suite:•Q.931 = ISDN call signalling•H.225.0 = RAS (registration/admissions/status) gatekeeping functions

+ Call signalling channel (CS), contains Q.931•H.245 = Control channel (CC), negotiation/capabilities, logical signalling,

maintenance •H.450.x = Supplementary services (SS), transfer, hold, park, msg wait, … incomplete!

H.225.0 RAS

H.225.0 CSH.245 CCH.450.x SS


119

Gatekeeper132.177.120.5

223-274910.0.0.5

3. Connect

4. Connect

1. Setup called: 5551234 caller: 9642749::10.0.0.5

5. TCS media: G.711/30ms, G.729/30ms

2. Setup called: 5551234::192.168.0.3 caller: 9642749

1, 5, 9, 13

4, 8, 12, 16

2, 6, 10, 14

3, 7, 11, 15

Gatekeeper Routed Call

Atlanta Zone (404)

223-4211192.168.0.3

6. TCS media: G.711/30ms, G.729/30ms

7. TCS media: G.729/20ms, G.723

8. TCS media: G.729/20ms, G.723

9. Open Channel G.729/30ms, 10.0.0.5:6400

10. Open Channel G.729/30ms, 10.0.0.5:6400

11. Open Channel G.729/20ms, 192.168.0.3:2300

12. Open Channel G.729/20ms, 192.168.0.3:2300

13. ACK 14. ACK 15. ACK 16. ACK


120

Gatekeeper132.177.120.5

223-274910.0.0.5

3. Setup called: 5551234 caller: 9642749::10.0.0.5

4. Connect

1. ARQ called: 5551234 caller: 9642749::10.0.0.5

5. TCS media: G.711/30ms, G.729/30ms

2. ACF called: 5551234::192.168.0.3

1

4, 6, 8, 10

2

3, 5, 7, 9

Gatekeeper Direct Call

Atlanta Zone (404)

223-4211192.168.0.3

6. TCS media: G.729/20ms, G.723

7. Open Channel G.729/30ms, 10.0.0.5:6400

8. Open Channel G.729/20ms, 192.168.0.3:2300

9. ACK

10. ACK


121

MEGACO/H.248, Softswitch Concepts


122

Master/Slave vs. Peer Comparison

•Lowest cost end device •Higher cost end deviceCost

•Lower performance “local” services•Sometimes higher

performance distributed services (e.g.. call control)

•Higher performance local services•High performance User

Interface

Performance

Feature deployment

•Update servers only•Services can come and go dynamically

•Update / download all end devices in network (yikes!)•Features more static per-device

Master/Slave (Thin Client) Peer (Thick Client)•Simple/dumb slave end device•Stimulus control, proxy in

network

•Smart/complex end device•Functional control, peer

interaction

Operation

Feature development

•Generic development tools•Shorter time to market for new features on a range of end devices•End device does not “get out of date” as quickly

•Device-specific development•Possibly shorter time to market

for new features on specific devices

•End device may need hardware upgrade over time

•MEGACO/H.248, MGCP •H.323, SIPProtocols


123

Megaco/H.248

Megaco

Audio Codec

G.711

G.723

G.729

Video Codec

H.261

H.263

RTP RTCP

LAN Interface

TCP

IP

UDP


124

Megaco/H.248 – Convoluted History

DSM-CCDSM-CC

DiameterDiameter

SGCPSGCP

IPDCIPDC

MGCP(proposal)

MGCP(proposal)

MDCP(proposal)

MDCP(proposal)

Megaco ProtocolMegaco Protocol

PacketCable NCS

PacketCable NCS

Megaco/H.248Megaco/H.248

Agreement reached between ITU SG16 and IETF Megaco to work together to create one standard (Summer 99)

ITU: H.GCPITU: H.GCP

MGCP proposal

PacketCable Network-based Call Signaling (NCS) based on earlier version of MGCP (March 99)

Megaco Protocol stream created, true consensus (March 99)

ITU SG-16 initiates gateway control project, H.GCP starting

from MDCP (May 99)

I-RFC 2705I-RFC 2705

MGCP released as Informational RFC (Oct 99)

WORLDSTANDARD

Industry Defacto

Std.

Non-Standard

Not fully accepted by Megaco WG, diverged (Spring 99)


125

Megaco Vs MGCP

Call Model Termination +Context +Topology P2P Single Media Single Media Conferencing P2P Multimedia Multimedia ConferencingTerminations Physical & Ephemeral & Muxing Template

Megaco/H.248

Event Packages (MGCP)Media Session Description SDPProtocol Encoding TextTransport UDP

MGCP

Command Grouping TransactionEvents Event BufferingEvent Packages (MGCP Packages + Additional Packages) National VariantsMedia Session Description SDP + H.245

Call Model Termination + Connection P2P Single Media Single Media ConferencingTerminations Physical & EphemeralCommand Grouping Ad hoc EmbeddingEvent Quarantine

Protocol Encoding Binary & TextTransport TCP + UDP +SCTPSecurity Authentication HeaderMGC Backup

Bold entries indicate additional features in Megaco vs. MGCP


126

Endpoint

(e.g.. H.323 Gateway,

Terminal, MCU)

Gateway Function

(e.g.. H.323 Gateway,

Terminal, MCU)

Call Agent

Media Gateway Controller

Signalling Gateway

PSTN,

ATM,

etc

trunks

lines

SS7 etc

Sigtran

AnalogMedia Gateway

PSTN trunkingMedia Gateway

PSTN lineMedia Gateway

IP PhoneMedia Gateway

Megaco Scope

Megaco Architecture Whirlwind Tour

Media Gateway Control Layer (MGC)• Contains all call control intelligence• Implements call level features (forward,

transfer, conference, hold, …)

Media Gateway Layer (MG)• Implements connections to/from IP cloud

(through RTP)• Implements or controls end device features

(including UI)• No knowledge of call level features

Signalling Gateway Layer (SG)• Interface to SS7 signalling etc• Not in Megaco scope (IETF Sigtran)

Media Gateway Control Protocol• Master / slave control of MGs by MGCs

–Connection control–Device control and configuration

• Orthogonal to call control protocols

Megaco Protocol

Call control (e.g.. H.323, SIP…)


127

Framework for H248/Megaco Protocol

IP (or ATM) Network

PBX/CO

PBX/CO

Media GW Controller

Media Gateway

Media Gateway• Connection and device control• No call processing, no call model• Service-independent• Cost effective

Devicecontrol

Devicecontrol

IP PhoneMedia

Gateway

Telephone/ResidentialMedia Gateway

PSTN trunkingMedia Gateway

PSTN lineMedia Gateway

Media GW Controller• Call processing and Service logic• Call routing• Inter-peer entity communication via

call control protocols (e.g. H.323, SIP, etc)

PBXMedia

Gateway


128

Megaco Framework The MGC and MGs form a virtual IP-based switch Looks like an H.323 Gateway to other H.323 devices, and a SIP Server to

other SIP devices RTP (the voice media itself) is still point-to-point

Media GW Controller

Media Gateways

Megaco/H.248

PSTN TrunkingMedia Gateway

PSTN LineMedia Gateway

Telephone/ResidentialMedia Gateway

Cable ModemMedia Gateway

Virtual SwitchSS7 Signalling

Gateway

Sigtrans

H.323H.323

Device

SIP

SIPDeviceRTP

RTP


129

Megaco call in action (optional)

ServiceChange: Restart

Reply: ServiceChangeReply: ServiceChange

Modify: Look for Off-Hook

MG1 MG2

Dial Tone,User Dials

Powered On

Powered On

ServiceChange: Restart

Modify: Look for Off-Hook

Ready ReadyReply: Modify Reply: Modify

MGC

Notify: Off-HookOff-HookReply: Notify

Modify: Dial Tone, Digit Map

Reply: Modify

Notify: number “19782886160”

Reply: Notify

Add: TDM to RTP, what codecs?

Reply: Add, codec G.729


130

Megaco call in action (continued)

Reply: Add

Add: TDM to RTP, ring phone

MG1 MG2

Open RTP Open RTPActive Call/End of Invite Request

Phone Rings

MGC

Modify: ip of MG2, ringback

Reply: ModifyHears Ring Off-HookNotify: Off-hook

Reply: Notify

Modify: stop ringStops Ring

Reply: ModifyModify: stop ringback, fullduplexReply: Modify

On-HookNotify: On-hookReply: Notify

Subtract:TDM and RTP

Reply: Subtract

Subtract: TDM and RTP

Reply: Subtract

Disconnect


131

Megaco/H.248 IP Phone Control

H.323MGC

Med

ia, L

CD, Sof

tkey C

ontro

l Media, LCD, Softkey Control



Voice (RTP)

Voice (RTP)

Voice

(RTP

)Voice (RTP)

In theory the RTP stream should go direct phone<->GW, but many today tandem through the MGC

In theory the RTP stream should go direct phone<->GW, but many today tandem through the MGCV

oic

e (R

TP

)

H.323 GWCisco’s Skinny,Nortel’s UNIStim,etc., are very similar protocols but they’re not interoperable

Cisco’s Skinny,Nortel’s UNIStim,etc., are very similar protocols but they’re not interoperable


132

Vendor Support for Standards

Source: Network World and Mier Communications - August, 2001

VoIP Protocol Support

30

64

54

56

81

66

32

57

77

22

11

30

42

30

40

17

26

73

0 10 20 30 40 50 60 70 80 90

Other

H.248 (Megaco)

MGCP (latest spec)

MGCP (orig. RFC2705)

SIP (Latest spec)

SIP (orig. RFC 2543)

H.323 other versions

H.323 V2

H.323 V1

Percentage of Vendors currentlly supporting the protocol

Percentage of vendors planning to add support within the next year


133

H.323 limitations

Gateway did a lot of things that were easily decomposed into functionally complete pieces Key insight from layering – separate functionally

complete pieces as far as possible. Quickly faced scaling problems

Call setup and control was a complex control plane operation

Media translation between a variety of networks Take-away point Build a distributed system that

acts as a single logical entity to the user


134

MGCP/H.248/Megaco

Media Gateway Controller(MGC)

Media Gateway

Media Gateway Controller(MGC)

SIP

Media GatewaySignaling Gateway Signaling Gateway

MGCP

Distributed entities acting in co-ordination

Connect to varietyof networks, home usersand other media receptorslike H.323 terminals etc

Interface tovariety of signaling mechanisms

User A

Separate signaling and voice planes, but

user unaware of it

Master/Slave

For examples of gateways see RFC 3435


135

Softswitch: Motivation

Class 5switch

Class 4switch

Class 5switch

UsersUsers

PSTN

Packet networks

Data

Voice

H.323 gateway

ISDN Switch

Class-4/5 switches bulky, expensive. Incentive to switch to cheaper easily managed IP

Initial gateway between PSTN and Internet was H.323. Gateway did signaling, call control, translation in one box. Not scalable.


136

What is a Softswitch?

A Softswitch is a device independent software platform designed to facilitate telecommunication services in an IP network

• A Softswitch controls the network

• At a high level, a Softswitch is responsible for:

• Protocol Conversion

• Control and synchronization of Media Gateways

• It’s an Architecture, NOT a box


137

The softswitch concept Build a distributed system that performs the functions of the Class-4/5

switches Use generic computing platforms to reduce cost, size and flexibility E.g., DSPs or other programmable architectures Software components to implement many of the switching tasks give

the “soft” part of “softswitch”

The MGC which does the call control and is the brain of the system is usually referred to as the softswitch or call agent

The gateways are dumb devices which do whatever MGC instructs them to do

MGC therefore does Call setup, state maintenance, tear-down

Megaco was an earlier non-standard framework which was later standardized jointly by ITU and IETF as MGCP


138

Softswitch: What’s the big deal? Unprecedented flexibility

Smaller offices can have just gateways, MGCs can be at some remote data center

Standards-based interactions drive down costs and offer wider architectural choices

Fast introduction of services and applications that can again be located remotely – only need MGCs to upgrade

New hosted-services solutions due to flexibility

Dramatic space savings Sometimes as much as 10 times smaller even with all the

components of the softswitch architecture


139

Softswitch Architecture

Application Server

Media Gateway

Signaling Gateway

Media GatewayController

PSTN/ End users

• Distributed functionality

• Open platforms

• Open interfaces enable new services

• Leverages the intelligence of endpoints

• Media agnostic


140

Softswitch - Media Gateway ControllerAn SS7 Enabled Media Gateway Controller integrates the functionality of new applications with the large installed based of legacy systems.

• Multiple controllers can collaborate on a single call

• May be distributed across the globe

• May or may not be collocated with SS7 Signaling Gateway

Application Server

Media Gateway

Signaling Gateway


PSTN/ End users


141

Softswitch - Media Gateway Controller Functions

• Connections (call setup and teardown)

• Events (detection and processing)

• Device management (gateway startup, shutdown, alerts)

Application Server

Media Gateway

Signaling Gateway


PSTN/ End users


142

Softswitch - Media GatewaysMedia Gateways provide interaction between audio in the network and software controlled applications

• Convert PSTN to IP packets

• Convert IP packets to PSTN

• In-band event detection and generation

• Compression (G.7xx,…)

• May be distributed across the globe

Application Server

Media Gateway

Signaling Gateway


PSTN/ End users


143

MGC and MG RolesMedia Gateway Controller

MGC’s allow intelligence to be distributed in the network

Basic call routing functions

Synchronization of Media Gateways

Protocol Conversion

Media Gateway

MG’s are purpose built specialist devices

Trunking gateways VoATM gateways Access gateways Circuit switches Network Access Servers


144

Softswitch - Signaling GatewaySignaling Gateways provide interaction between the SS7 network and Media Gateway Controllers.

• Convert SS7 to IP packets

• Convert IP to SS7 packets

• Signaling transport (SS7, SIP-T, Q.931…)

• Extremely secure

• Extremely fault tolerant

Application Server

Media Gateway

Signaling Gateway


PSTN/ End users


145

Softswitch – Application ServerApplication Servers(AS) provide the new services that are the real “value-add” for Softswitches.

• Many core features are part of the MGC

• Allows new features to be developed by third parties

Application Server

Media Gateway

Signaling Gateway


PSTN/ End users


146

Softswitch – Application ServerApplication Servers(AS) Can be broken apart and distributed in the network

Feature Server

Policy Server

Directory Server

Media Server Management Server

LDAP

Corba

SIP,Parlay,JAIN

Connectivity Server

SIP

COPSNetwork Elements

Corba


147

Softswitch Architecture – The protocolsApplication

Server

Media Gateway

Signaling Gateway


PSTN/ End users

SIP, Parlay, Jain

Sigtran w/SCTP

H.248,MGCP


148

Softswitch Architecture – Interdomain protocols

Application Server

Media Gateway

Signaling Gateway


PSTN/ End users

SIP, Parlay, Jain

Sigtran

H.248,MGCP

Application Server

Media Gateway

Signaling Gateway


PSTN/ End users

SIP-T,BICC

RTP

Application specific


149

SIP vs MEGACO: Summary


150

SIP vs MEGACO (contd)


151

VoIP Signaling Model: Summary

End-system: SIP signaling (beat out H.323) PSTN gateway, with interfaces looking into PSTN

and interfaces looking into VoIP networks Media Gateway Controller (MGC): “intelligent”

endpoint: supervises call services end-end Media Gateway (MG): interface to the IP network or

PSTN: “simple” endpoint instructed by MGC MEGACO: MG MGC interaction protocol;

ITU (H.248) and IETF (RFC 3525) standard Replaces proprietary APIs and RFC 3435 (MGCP)


152

Speech Coding and Speech Coders for VoIP


153

Taxonomy of Speech CodersSpeech Coders

Waveform Coders Source Coders

Time Domain: PCM, ADPCM

Frequency Domain: e.g. Sub-band coder,Adaptive transform coder

Linear Predictive Coder

Vocoder

Waveform coders: attempts to preserve the signal waveform not speech specific (I.e. general A-to-D conv)

PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbpsVocoders:

Analyse speech, extract and transmit model parameters Use model parameters to synthesize speech LPC-10: 2.4 kbps

Hybrids: Combine best of both… Eg: CELP (used in GSM)


154

Speech Quality of Various Coders


155

Speech Quality (Contd)


156

Actual Bandwidth Used

Framesizein ms

PacketIn bytes

+ RTP+UDP+IPin bytes

LANframe inbytes

T-LANkbps

WANkbps

G.711(64 kbps)

102030

80160240

120200280

146226306

116.890.481.5

96.080.074.6

G.729A/G.729( 8 kbps)

102030

102030

506070

768696

60.834.425.6

40.024.018.6

G.723.1(5.3 kbps)

30 20 60 86 22.9 16.0

G.723.1(6.3 kbps)

30 24 64 90 24.0 17.0

Note: (1) 26-bytes Ethernet overhead was removed for WAN calculation. (2) No backbone protocol overhead was used for WAN bandwidth. (3) This is per voice direction, so multiply by 2 if on a shared (half-duplex) media

(4) No Ethernet Interframe Gap was included (another 12 bytes)


157

Applications of Speech Coding

Telephony, PBX Wireless/Cellular Telephony Internet Telephony Speech Storage (Automated call-centers) High-Fidelity recordings/voice Speech Analysis/Synthesis Text-to-speech (machine generated speech)


158

Pulse Amplitude Modulation (PAM)


159

Pulse Code Modulation (PCM)

* PCM = PAM + quantization


160

Companded PCM

•Small quantization intervals to small samples and large intervals for large samples• Excellent quality for BOTH voice and data• Moderate data rate (64 kbps)• Moderate cost: used in T1 lines etc


161

How it works for T1 Lines

• Companding blocks are shared by all 16 channels


162

Recall: Taxonomy of Speech CodersSpeech Coders

Waveform Coders Source Coders

Time Domain: PCM, ADPCM

Frequency Domain: e.g. Sub-band coder,Adaptive transform coder

Linear Predictive Coder

Vocoder

Waveform coders: attempts to preserve the signal waveform not speech specific.

PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbpsVocoders:

Analyse speech, extract and transmit model parameters Use model parameters to synthesize speech LPC-10: 2.4 kbps

Hybrids: Combine best of both… Eg: CELP


163

Vocoders

Encode only perceptually important aspects of speech w/ fewer bits than waveform coders: eg: power spectrum vs time-domain accuracy


164

LPC Analysis/Synthesis


165

Speech Generation in LPC


166

CELP Encoder


167

Example: GSM Digital Speech Coding

PCM: 64kbps too wasteful for wireless

Regular Pulse Excited -- Linear Predictive Coder (RPE--LPC) with a Long Term Predictor loop.

Subjective speech quality and complexity (related to cost, processing delay, and power)

Information from previous samples used to predict the current sample: linear function.

The coefficients, plus an encoded form of the residual (predicted - actual sample), represent the signal.

20 millisecond samples: each encoded as 260 bits =>13 kbps (Full-Rate coding).


168

Standard Algorithm Bit Rate (Kbit/s) Codec Induced Delay (msecs)

Resultant Voice Quality

G.711 PCM 56, 64 <<1 Excellent G.723.1 MPE/ACELP 5.3, 6.3 67-97 Fair(5.3), Good(6.3) G.728 LD-CELP 16 <<2 Good G.729 CS-ACELP 8 25-35 Good G.722 Sub-band

ADPCM 64 5-10 Good-Excellent (it’s

wideband) G.726 ADPCM 16, 24, 32, 40 <<1 Fair(24), Good(40) GSM-EF ACELP 12.2 40 Good

Codecs: Quality Measures

Only G.711, G.723.1, and G.729 are popular (because they are mandatory for several specs)

G.711 is the best (obviously), but G.729 isn’t much worse G.723.1 is HORRIBLE


169

PreambleDestination

AddressData Pad Checksum

SourceAddress

Inter-framegap

Start of framedelimiter

Length orEthertype

12 7 1 6 6 2 0-1500 0-46 4

Ethernet Frame

DestinationAddress

SourceAddress

HeaderChecksum

DataFlags &

Frag OffsetTotal

LengthPacket

IDOptions(if any)

1 1 2 2 2 1 1 2 4 4 0-40 0-1480

IP packet

Version &header length

TOS TTL

Protocol

SourcePort Number

DestinationPort Number

UDP length UDP checksum

2 2 2 2 0-1472

UDP datagram

Version,flags & CC

SequenceNumber

Timestamp

1 1 2 4 4 0-60 0-1460


Source IDPayload

TypeCSRC ID(if any)

Codec Data

Data

Packet Encapsulation


170

42668 120IP into Ethernet

IP PayloadPreamble

Destination

Source

Type CRC.

80 byte voice bundles

RTP Frame 12RTP Header Voice Payload

80

2222 80UDP Datagram

Voice Payload

Destination

Source

LengthChecksum

Destination

12

RTP Header

1 12 2120IP into Frame Relay Flag

FlagAddress Frame Check

IP Payload

4412 80IP Packet Header

SourceVoice PayloadUDP Header RTP Header

8 12

5 48 5 24+IP into ATMIP Payload

5 48 +IP Payload IP Payload

Header Header Header

16

Padding Trailer

8

G.711 (10ms) Clear Channel Voice


171


IP PayloadPreamble

Destination

Source

Type CRC.



30

2222 30UDP Datagram

Voice Payload

Destination

Source

LengthChecksum

Destination

12

RTP Header



IP Payload



8 12

5 48 5 22IP into ATMIP Payload

+IP Payload

Header Header

18

Padding Trailer

8



172


IP PayloadPreamble

Destination

Source

Type CRC.



20

2222 20UDP Datagram

Voice Payload

Destination

Source

LengthChecksum

Destination

12

RTP Header



IP Payload



8 12

5 48 5 12IP into ATMIP Payload

+IP Payload

Header Header

28

Padding Trailer

8



173

20-24 byte voice bundles


20-24

42668 60-64IP into Ethernet

IP PayloadPreamble

Destination

Source

Type CRC.

2222 20-24UDP Datagram

Voice Payload

Destination

Source

LengthChecksum

Destination

12

RTP Header

1 12 260-64IP into Frame Relay Flag


IP Payload

4412 20-24IP Packet Header


8 12

5 48 5 12-16IP into ATMIP Payload

+IP Payload

Header Header

28-24

Padding Trailer

8

G.723.1 (30ms) Clear Channel Voice


174

Coding Technology Side-effects

Coded VoIP is NOT the same as a telephone line (I.e. it is not a content-neutral “carrier”): Without special support, you cannot send “fax” or “modem

traffic” over VoIP The “carrier” is now IP (or some data-transport protocol

like frame-relay or ATM) The same is true for 3G or GSM telephony Why? Voice is encoded and the encoding works only for

voice! (it is no longer a 64 kbps bit stream) Fax support: Fax Passthru, T.38 fax Relay


175

Voice Quality: Loss Tolerance Voice codecs are unevenly tolerant of packet loss,

but loss above 2 to 5 percent will have a perceptible effect on quality.

Losses also associated with higher jitter 1-way delay > 150 milliseconds, => trouble Jitter buffer (major component of delay budget) Capacity reservations & priority for key packets:

setup through RSVP Priority: using TOS bits: 8 levels of precedence

Carrier networks use some combination of: MPLS (traffic engineering, stable routing) and Diff-serv (expedited forwarding) to provide

superior service for VoIP


176

VoIP QoS Myths Packet voice=> voice could take multiple paths or failover.

But it usually does not…

VoIP is sensitive to routing failures or congestion in paths OSPF and BGP convergence times too bad for VoIP:

SONET and (now) MPLS much better

However, FEC packets for VoIP can be sent on a separate path or on the same path: hedge against performance fluctuations (eg: congestion) on

the primary path, but limited hedge against failure of the primary path.


177

Voice codecs: Summary G.711

uncompressed PCM audio stream 8ks/s of 8 bit values = 64kbps packet “sizes” = 10, 20, 30 and 60ms

G.722 - Wideband (7kHz) G.726

ADPCM - 10,20,30,60ms - 32kbps G.723.1

MLQ - 30ms - 5.3 or 6.3kbps Silence suppression

G.729 CS-ACELP - 10, 20, 30ms - 8kbps Annex B adds silence suppression


178

Recap: Speech Quality of Various Coders


179

Miscl: Other standards, ENUM, E-911, Presence etc


180

Sigtrans (Signaling Transport) Signalling transport protocol and adaptation layers for SG to

MGC communication, and for SG to SG communication Signalling Gateways can be stand-alone or co-located with an

MGC

Media GW Controller

Trunk Gateway

Megaco/H.248

Virtual Switch

Sigtrans SIP, H.323

Media Gateway

Signalling Gateway

PBX

D-channel

B-channels

PRI

Virtual Switch

RTP

Sig

tran

s

Signaling Gateway

Signaling Gateway

SS7

CO

Sigtrans

Sig

tran

s


181

SCTP (Stream Control Transmission Protocol)

Sigtrans needs to carry SS7 Needed a reliable transport mechanism (like TCP) without the overhead

of a connection-oriented protocol SCTP created: like UDP, but with acknowledgment, fragmentation, and

congestion-avoidance This has much broader use than just carrying SS7: it’s being looked at

for SIP, RTP, T.38, and more...

6 - Presentation5 - Session User Adaptation Modules 4 - Transport SCTP3 - Network IP2 - Link MLPPP / FR / ATM1 - Physical Ethernet / SONET/Serial


182

(1) SS7 Signaling Using IP Transport

Applications

MTP2

IP

SCTP

SSP

STP

MTP2

MTP3

SCCP

TCAP

ISUP

Applications

SSP

IP

SCTP

The IETF M2UAMTP2-User Adaptation Layer

from the Sigtran WG

M2UA

MTP3*

M2UA

MTP3

SCCP

TCAP

ISUP


183

(2) SS7 / IP Interworking

CallProcessingApplication

MTP2IP

SCTP

SSP MGC

SS7 SG

M3UA

Nodal Inter-working Function

CallProcessingApplication

MTP3

MTP2

MTP3

ISUP

IP

SCTP

M3UA

ISUP

The IETF M3UAMTP3-User Adaptation Layer

from the Sigtran WG


184

BICC (Bearer Independent Call Control) Offers a migration path from SS7/TDM to packet-based

voice Defines Interface Serving Node for Bearer, Bearer Control,

and Call Serving Functions Specifies Transit Serving Nodes to change bearer types,

and Gateway Serving Node to transit operators

PSTN PSTN

Class 4 Switch Class 4 Switch

Data Network

BICC ISN BICC ISN

BICCISUP ISUP

TDMTDM


185

VPIM (Voice Profile for Internet Mail) Uses SMTP to send/receive voice/faxmail messages Attaches messages as wav/mpeg/tiff files in MIME Useful for transferring across voicemail systems Adds more useful info: vcard, signature, multiple addresses POP3 still used to download voicemail to your favorite email

client (Outlook, Eudora, Pine, etc.)

PBXUnified

Messaging System

Unified Messaging

System

SIP/H.323

SIPDevice

VPIMEmail

Browser

POP3

Plain Phone


186

TRIP – Telephony Routing over IP

TRIP is a protocol for advertising the reachability of telephony destinations between location servers, and for advertising attributes of the routes to those destinations.

Can serve as a routing protocol for any signaling protocol TRIP is used to distribute telephony routing information

between telephony administrative domains. TRIP is essentially BGP for phone numbers and the

protocol is actually based on BGP-4


187

Proxy server

parliament.uk

[email protected]

Location Server

ge

org

e.w

.bu

sh d

ch

en

ey

@w

h

3. SIP/2.0 200 ok From: sip:dcheney@wh

4. SIP/2.0 100 OK From: sip:[email protected]

1. INVITE sip:[email protected] SIP/2.0 From: sip:[email protected]

5. ACK sip:[email protected] SIP/2.0 From: sip:[email protected]

6. ACK sip:dcheney@wh SIP/2.0 From: sip:[email protected]

2. INVITE sip:dcheney@wh SIP/2.0 From: sip:[email protected]

1 & 5

4

2 & 6

3

Midcom (Middlebox Communication)

us.gov

dcheney@wh

3.5 Midcom Protocol

Firewall


188

Mediation and Billing

Current State

Non real time Non-scalable Limited functionality No revenue assurance capabilities Proprietary CDR formats No OSS functionality (fraud, churn, etc.) Mainly stand alone systems (no integration with the

legacy systems)


189

Call Detail Records

• To be able to run reports and bill, Call Detail Records (CDRs) must be recorded for each call:

With VoIP far more detail is necessary: Packets transmitted Packets lost Jitter Delay Call Control / Gateway used Codec used …

Time Reason From To Duration

Details

16:45 Call req. 5551212 6663434 01:45 Normal disc.


190

Mediation and Billing Requirements Complete call details including

Call descriptors caller ID, called #, time, length, disconnect reason, QoS requested, etc.,

Complete network QoS information (dropped packets, trunk failure, etc.)

Complete application level QoS (dropped frames, disconnect reason, CODEC type, etc.)

Carrier-grade solution Scalable Large number of calls/sec Cover large, distributed networks

Real Time Revenue Assurance

99.999% accuracy Audit capabilities Highly available

Support of standards Integration with other OSS/BSS systems (fraud, churn, etc)

Fault tolerantLocal cache Roll back


191

IPDR – IP Data Records

The purpose of the IPDR initiative is to define the essential elements of data exchange between network elements, operation support systems and business support systems. Specific goals include:

Define an open, flexible record format (the IPDR record) for exchanging usage information.

Define essential parameters for any IP transaction. Provide an extension mechanism so network

elements and support systems exchange optional usage metrics for a particular service.


192

ENUM vs DNS DNS (or internet) names: interpreted right to left:

Eg: www.rpi.edu Telephone numbers: interpreted left to right:

Eg: +1 518 276 8979 ENUM: (RFC 3761)

telephone numbers written DNS-style, Rooted at the domain e164.arpa. So, 1.212.543.6789 becomes 9.8.7.6.3.4.5.2.1.2.1.e164.arpa. When queried, DNS can return an IP address for the telephone

number, or it can return a rule for re-formatting the original number For example, rules can be returned to rewrite 1.212.543.6789 as

sip:[email protected], sip:[email protected].


193

Continuity of Telephone Svcs in VoIP

A number of basic features remain same: Phone looks and behaves like a phone DTMF (touch-tone) features: mid-call signaling E.911 will provide 911 location services Bearer (“data-plane”) is separated from signaling

(“control-plane”) and is handled differently But, unlike telephony, it is multiplexed on the

same network Interfaces smoothly with internet applications: IM,

Web, email…


194

E911 - Requirements

911 Services

Power stays on when building power fails

Need callers phone number and location

Services must be modified during a 911 callDisable call-waitingDisable three-party callsCaller cannot hangup and place another call


195

E911 – VoIP Enhancements

VoIP has the potential of enhancing E911 functionality Multimedia communication

Audio – emulate existing servicesVideo – images and/or biometrics to/from emergency

techniciansText – for hearing impaired

Call setup could contain medical backgroundCan be locally maintained, does not a master database

Calls can easily be forwarded or transferredFast call setup times

PSAP could easily be deployed or relocated anywhere Internet access is available.


196

E911 – Using DNS to convey location Based on network device name

pigface 192.168.200.20

GL S3.US.95401.4500 “110 Stony Point Rd.,Santa Rosa CA” Based on Geographic location (longitude/latitude)

pigface 192.168.200.20

GPOS -38.43954 122.72821 10.0 Binary (includes precision indicator)

pigface 192.168.200.20

LOC 23 45 32 N 89 23 18 W –24m 30m

Issues Only works if mapping between device and location is correct. Not secure/private


197

Invisible Internet Telephony

VoIP technology will appear in . . . Internet appliances home security cameras, web cams 3G mobile terminals fire alarms chat/IM tools interactive multiplayer games


198

VoIP Reliability & Manageability

Reliability: PSTN benchmarks… Work all the time, except for maintenance windows Faults: network, hardware, software Duplicated systems: no upgrade downtime Monitors, automatic failovers

Manageability: accurate and flexible billing systems, error reporting and resolution, call tracing, adds/moves/changes, Lack of network state (IP model) makes this

difficult => mediated calls (eg: softswitch etc reinstate some of this…)


199

IPtel for appliances: “Presence”


200

VoIP Standards (Enterprise View)

3rd PartyCall Servers &Gatekeepers

RTP

H.323 annex G,

SIP

H.323Gateway

SIPGateway Stimulus

Terminals

ThickTerminal

s

EnterpriseCall

Server

SIP H.248, Stimulus

H.323H.323

RTP

RTP

RTP

IP-enabledPBX/KS

H.248,Stimulus

SIP, H.323

H.323, SIP, Q.Sig

SIP, H.323

RTP

RTP


201

VoIP Standards (Carrier View)

3rd PartyCall Agents &Gatekeepers

RTP

H.323, SIP-TBICC

Application/Media Server

SIPGateway

MegacoGateway

Softswitch/ Call Agent/

MGC

SIP Megaco/ H.248 MGCP

RTP

RTP

RTP

Signalling(SS7)

Gateway

SIP

Sigtrans, Q.BICC

MGCPGateway


202

VoIP Summary: Big Picture

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