Vo ip sip

© 2007 Cisco Systems, Inc. All rights reserved. Cisco ConfidentialPresentation_ID 1

VoIP & SIP Signaling

Hussam El Kebbi

Presentation_ID 2© 2007 Cisco Systems, Inc. All rights reserved. Cisco Confidential

Overview VoIP Architecture What is VoIP ? Benefits of VoIP Components of a VoIP Network Quiz VoIP Telephone Call Overview on a VoIP Connection

Analog/Digital Interfaces

Steps of Conversion Transport Layer QoS in VoIP/Solutions for QoS Issues Calculating Bandwidth Requirements for VoIP Cisco IOS Configurations for VoIP Quiz


SIP Architecture What is SIP? SIP Capabilities SIP URI / Components Quiz

SIP Message Format SIP Message Request/Response Fields Header Fields Quiz

SIP Call Flow Using Proxy Server/Using Multiple Servers How are the Codecs Negotiated / SDP Information SIP Security Mechanisms QUIZ

Overview


VoIP Architecture


VoIP Architecture

Voice over IP (VoIP) defines a way to carry voice calls over an IP network including the digitization and packetization of the voice streams

What is VoIP ?

http://www.cisco.com/en/US/tech/tk652/tk701/tsd_technology_support_protocol_home.html


Benefits of VoIP

More efficient use of bandwidth and equipment

Reduce operating costs

Consolidated network expenses

Improved employee productivity

Access to new communication devices

VoIP Architecture

CCNP: Optimizing Converged Networks v5.0NT, Chapter 2


Components of a VoIP NetworkThese are the most common elements in VoIP networks:

VoIP Architecture



Quiz

Define VoIP, and list two of its benefits ?

List three of VoIP network components ?

VoIP Architecture


VoIP Telephone Call


To setup a VoIP communication we need: Convert analog voice to digital signals (bits) Now the bits have to be compressed in a good format for

transmission Insert our voice packets in data packets using a real-time protocol

(typically RTP over UDP over IP) We need a signaling protocol to call users (SIP - H.323) At Receiving we have to disassemble packets, extract data, then

convert them to analog voice signals All that must be done in a real time fashion cause we cannot waiting

for too long for a vocal answer! (QoS )

Overview on a VoIP connection

http://tldp.org/HOWTO/VoIP-HOWTO-4.html

VoIP Telephone Call

http://tldp.org/HOWTO/VoIP-HOWTO-4.html


Foreign Exchange Station (FXS)

Foreign Exchange Office (FXO)

Ear and Mouth (E&M)

Legacy Analog Interfaces in VoIP Networks

Digital Interfaces

Analog Interfaces

VoIP Telephone Call


Click me


Converting Analog Signals to Digital Signals Sample the analog signal (Sampling)

Quantize sample into a binary expression (Quantization)

Compress the samples to reduce bandwidth

Converting Digital Signals to Analog Signals Decompress the samples

Decode the samples into voltage amplitudes

Reconstruct the analog signal


VoIP Telephone CallSteps of Conversion

Click me


To calculate the total bandwidth, find the total packet size, including all the headers plus payload and divide by the payload size. Multiply the result by the nominal bandwidth for the codec. The result is the total bandwidth requirement.

VoIP Telephone Call

Calculating Bandwidth Requirements for VoIP


You can calculate the bandwidth using the Voice Codec Bandwidth Calculator at http://tools.cisco.com/Support/VBC/do/CodecCalc1.do.


Real-time applications such as voice and video require a guaranteed connection with consistent and predictable delay characteristics.

IP does not guarantee reliability, flow control, error detection, or error correction TCP offers both connection-oriented and reliable transmission

- Handles sequencing and error detection to ensure that the destination application receives a reliable stream of data UDP, like IP, is a connectionless protocol.

- Routes data to its correct destination port but does not attempt to perform any sequencing or to ensure data reliability

Transport Layer

VoIP Telephone Call



RTP solves the problem enabling the receiver to put the packets back into the correct order and not wait too long for packets that have either lost their way or are taking too long to arrive

- RTP transports the digitized samples of real-time information

- The packets can be correctly reordered

- The packets can have appropriate delays inserted between packets

Transport Layer

VoIP Telephone Call

VoIP doesn't use TCP because it is too heavy for real time applications



Latency: Delay for packet delivery

Jitter: Variations in delay of packet delivery

Packet loss: Too much traffic in the network causes the network to drop packets

Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in bursts

QoS (Quality of Service) is a major issue in VOIP implementations, things to consider are:

VoIP Telephone Call

QoS in VoIP

http://www.voip-info.org/wiki/view/QoS



Solution for QoS issues

Resource reservation : Make sure that the VoIP call has the bandwidth needed allocated from point to point before the conversation takes place.

Prioritization: Here, the end point suggest a priority on the packets and each router decides if it will honour this request or not.

Network Traffic Tuning: Boxes you can add to a network to manage bandwidth usage and create QOS even if the other network devices don't support it.

VoIP Telephone Call




Cisco IOS Configurations for VoIP

http://www.cisco.com/en/US/products/hw/routers/ps221/products_configuration_guide_chapter09186a008007c9bc.html


Quiz Which three components should be taken into

consideration when calculating the voice bandwidth needed to set up a call on a VoIP network? (Choose three)

1. Voice payload size

2. RTP, UDP, and IP headers

3. Layer 2 encapsulation

4. Low latency queuing (LLQ)

5. Classification and marking of the voice traffic

6. Call Admission Control enabled on the network

VoIP Telephone Call


Does VoIP use TCP as transport protocol?Why?

VoIP Telephone CallQuiz


SIP Architecture


SIP Architecture

The Internet Engineering Task Force's (IETF's) standard for multimedia conferencing over IP

A signaling protocol used to create, manage and terminate sessions in an IP based network.

A client/server protocol, which is similar to HTTP Influencing the marketplace, a growing number of IP

Telephony Service Providers (ITSP)/ cellular phone providers, Microsoft real-time communication platforms, and Cisco applications are based on SIP

Session Initiation Protocol (SIP) is :What is SIP?


Presence, Instant Messaging and Voice


Determine the location of the target end point

Determine the media capabilities of the target end point—Via Session Description Protocol (SDP)

Determine the availability of the target end point Establish a session between the originating and target end point

Handle the transfer and termination of calls

SIP Capabilities :

SIP Architecture

Click me


A user of an online service An appearance on a multiline phone A mailbox on a messaging system A telephone number at a gateway service

SIP URIs have a format based on e-mail address formats, namely user@domain. There are two common schemes. An ordinary SIP URI is of the form: sip:[email protected]

The URI may also include a password, port number, and related parameters.

SIP ArchitectureSIP Universal Resource Indicators

http://www.cisco.com/web/about/ac123/ac147/archived_issues/ipj_6-1/sip.html

A resource within a SIP configuration is identified by a URI. Examples of communications resources include the following:



User agent client (UAC)

User agent server (UAS)

SIP Components

Functional Components : Proxy server : Perform call routing, authentication, authorization, address resolution, and loop detection

Redirect server : UAs and proxy servers can contact a redirect server to find the location of an end point

Registrar : Processes requests from UACs for registration of their current location

SIP Architecture

SIP is a peer-to-peer protocol, can function in one of the following roles:

Click me

SIP User Agents


Quiz

What is SIP?

Name 3 Funtional Components of SIP Architecture?

SIP Architecture


SIP Message Format


SIP Message Format

SIP is a simple, ASCII text-based protocol that uses requests and responses to establish communication among the various components in the network

generic-message = start-line ;start-line = Request-Line / Status-Line*message-headerCRLF ; carriage-return line-feed sequence [ message-body ]

SIP Message

http://www.tech-invite.com/Ti-sip-abnf.html



Status-Line = SIP-Version SP Status code SP Reason-phrase CRLF

SIP version : The SIP version being used.

Status-code : A 3-digit integer result code of the attempt to understand and satisfy the request.

Reason-phrase : A textual description of the status code.

SIP Message Format

Response Fields

Request FieldsRequest-Line = Method SP Request-URI SP SIP-Version CRLF Method : Register, Invite, Ack, Cancel, Bye, and Options Request-URI : It indicates the user or service to which this request is being addressed SIP version : Is case-insensitive but implementations must send upper case


Provisional (1xx): The request was received and is being processed.

Success (2xx): The action was successfully received, understood, and accepted.

Redirection (3xx): Further action needs to be taken in order to complete the request.

Client Error (4xx): The request contains bad syntax or cannot be fulfilled at this server.

Server Error (5xx): The server failed to fulfill an apparently valid request.

Global Failure (6xx): The request cannot be fulfilled at any server.

SIP Message FormatResponseThe SIP response types defined in RFC 3261 are in the following categories:

www.cisco.com/web/about/ac123/ac147/archived_issues/ipj_6-1/sip.html



Header Fields

To : Specifies the desired "logical" recipient of the request

From : Indicates the logical identity of the initiator of the request

CSeq : Provide a means to uniquely identify transactions

Call-ID : Acts as a unique identifier to group together a series of messages

Contact : Provides a URI whose meaning depends on the type of request or response it is in.

Via : Indicates the path taken by the request so far and indicates the path that should be followed in routing responses

SIP Message Format

A valid SIP request formulated by a UAC MUST, at a minimum, contain following header fields:

http://www.networksorcery.com/enp/protocol/sip.htm




INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 12.26.17.91:5060Max-Forwards: 70To: Bob <sip:[email protected]: Alice <sip:[email protected];tag=1928301774Call-ID: [email protected]: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 142

SIP Message Format

Request Example




Response Example SIP/2.0 200 OK

Via: SIP/2.0/UDP server10.biloxi.comVia: SIP/2.0/UDP bigbox3.site3.atlanta.comVia: SIP/2.0/UDP 12.26.17.91:5060To: Bob <sip:[email protected];tag=a6c85cfFrom: Alice <sip:[email protected];tag=1928301774Call-ID: [email protected]: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 131

SIP Message Format




Quiz

SIP Message code basis are ?

Binary

ASCII

What are the Fields of SIP Message ?

SIP Message Format


SIP Call Flow


SIP Call FlowFigure 1, Using proxy server


Figure 2, Using multiple servers

SIP Call Flow


How are the Codecs negotiated? SDP is the protocol used by the UAs to tell each other what codecs they

support. SDP is embedded into the SIP Messages.SDP is intended for describing multimedia sessions for the purposes

of session announcement, session invitation, and other forms of multimedia session initiation

SDP, defined in RFC 2327, describes the content of sessions, including telephony, Internet radio, and multimedia applications

SIP Call Flow

Session Descripton Protocol (SDP)

v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.6.2.10 s=SIP Call c=IN IP4 10.6.2.10 t=0 0 m=audio 24580 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/800

http://www.ietf.org/rfc/rfc2327.txt


Media streams: A session can include multiple streams of differing content. SDP currently defines audio, video, data, control, and application as stream types.

Addresses: SDP indicates the destination addresses, which may be a multicast address, for a media stream.

Ports: For each stream, the UDP port numbers for sending and receiving are specified.

Payload types: For each media stream type in use (for example, telephony), the payload type indicates the media formats that can be used during the session.

Start and stop times: These apply to broadcast sessions, for example, a television or radio program. The start, stop, and repeat times of the session are indicated.

Originator: For broadcast sessions, the originator is specified, with contact information. This may be useful if a receiver encounters technical difficulties.


SIP Call FlowSDP Information :



Why SDP is used ?

SIP Call FlowQuiz


SIP Security Internet Telephony uses a variety of signaling protocols,

such as H.323, SIP, MGCP and MEGACO, for initiating VOIP calls.

SIP, like other Internet Protocols, is vulnerable to known Internet attacks.

VOIP suffers from all known attacks associated with any Internet application or subsystem

SIP Security Mechanisms

Saverio Niccolini, Ph. D.Research Staff Member @ Network Laboratories NEC Europe Ltd


SIP Client-A drops the call just initiated



DoS Attack


Call Hijack



Threats


Identity TheftSIP Security Mechanisms



Registration and call signaling/media should be authenticated

End-to-end

- Digest authentication (challenge - response)

- S/MIME Hop-by-hop

- TLS, IPsec

- SIPS



Ways to ensure signaling security in SIP: HTTP Digest: prone to eavesdropping, replay, and MiTM attacks.

Provides authentication only.

TLS: Hop-by-hop SIP transport security; not end-to-end! Provides confidentiality, authentication, encryption.

S/MIME : End-to-end signaling and body security. Provides confidentiality, authentication, encryption.

IPSec: Layer 3 security. Provides confidentiality and encryption.



Quiz

Which are Hop-by-hop / End-to-end SIP transport security?

TLS

IPSEC

HTTP Digest

S/MIMe End-to-end



Presence, Instant Messaging and Voice

Thank You Hussam El Kebbi



Presentation_ID 52© 2007 Cisco Systems, Inc. All rights reserved. Cisco Confidentialhttp://www.cs.columbia.edu/IRT/sipc/doc/html/images/monitor.png

http://www.cs.columbia.edu/IRT/sipc/doc/html/images/monitor.png

Presentation_ID 53© 2007 Cisco Systems, Inc. All rights reserved. Cisco ConfidentialCCNP: Optimizing Converged Networks v5.0NT, Chapter 2

Presentation_ID 54© 2007 Cisco Systems, Inc. All rights reserved. Cisco ConfidentialCisco Interfaces

BRI

T1

EI

Presentation_ID 55© 2007 Cisco Systems, Inc. All rights reserved. Cisco ConfidentialCCNP: Optimizing Converged Networks v5.0NT, Chapter 2

Presentation_ID 56© 2007 Cisco Systems, Inc. All rights reserved. Cisco Confidentialhttp://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/bgsipov.pdf

http://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/bgsipov.pdf

Presentation_ID 57© 2007 Cisco Systems, Inc. All rights reserved. Cisco Confidentialhttp://www.tech-invite.com/Ti-sip-abnf.html



SIP Components / Servers / Services

SIP User Agents

Registrar RedirectLocation

Database

SIP Proxy

SIP Servers /Services

REGISTER“Here I am”

INVITE“I want to talk

to another UA”

Proxied INVITE“I’ll handle it for

you”

“Where is this name/phone#?”

3xx Redirection“They moved,

try this address”

SIP User Agents SIP-GW

http://www.cisco.com/




Date post:	07-Nov-2014
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