VoIP Signaling Protocols

A signaling protocol is a common language spoken by telephones and call-management servers, the PSTN, and legacy PBX systems as they communicate to set up, monitor, and tear down calls.

The Voice over IP (VoIP) technology family provides several signaling protocols. Asterisk support most of them but a few will be discussed here:

H.323SIP IAX

H.323H.323 is an International Telecommunications Union

Telecommunications Standardization Sector (ITU-T) specification for transmitting multimedia traffic, including video and voice, over an IP network

H.323 ProtocolsFeature Protocol

Call Signalling H.225

Media Control H.245

Audio Codecs G.711, G.722, G.723, G.728, G.729

Video Codecs H.261, H.263

Data Sharing T.120

Media Transport RTP/RTCP

H.323 Elements

H.323 elements include terminals, gateways, gatekeepers and Multipoint Control Units (MCUs).

TerminalsAlso known as endpoints, terminals provide point-to-point and

multipoint conferencing for audio, video and dataGateways Gateways are used to connect between Switched Circuit

Network (SCN) endpoints and H.323 endpoints. Gateways are only needed when an H.323 endpoint needs to interconnect to a different network

GatekeeperGatekeepers provides pre-call and call-level control services

to H.323 endpoints.

H.323 elements

Multipoint Controller (MC)A Multipoint Controller supports conferencing between three

or more endpoints. A Multipoint Processor (MP) receives audio, video and data streams, and then redistributes those streams to the endpoints in a multipoint conference

The H.323 Call-Signaling Process

There are five general steps in the H.323 signaling process: setup/teardown, capabilities negotiation, open media channel, perform call, and release.

 Setup/TeardownTo initiate an H.323 call, H.225 is required for the setup process. The following are the most commonly used signaling messages :Setup: A forward message sent by a calling entity in an attempt

to establish a connection with the called entityProceeding: A backward message sent from the called entity to

the calling entity to inform that call establishment procedures were initiated

The H.323 Call-Signaling ProcessAlerting: A backward message sent from the called entity to inform that

called party ringing was initiatedConnect: A backward message sent from the called entity to the calling

entity that the called party answered the call. The connect message can contain the transport UDP/IP address for H.245 control signaling

Release: sent by endpoint initiating disconnect

Capabilities NegotiationAfter setup, H.245 is enlisted to negotiate the call’s application

requirements H.245 determines:-Which kind of application media each terminal can support: audio,

video.-Which codecs each terminal is capable of and which it may prefer-How the media channel will be structured, and which packet interval

will be used-Which terminal will be the master and which will be the slave for the

duration of the call. Master and slave roles distinguish the client/server role assumptions for future signals during the call and are a protocol formality

The H.323 Call-Signaling ProcessOpen Media Channel Once capabilities negotiation has succeeded, RTP Control

Protocol (RTCP) establishes a UDP socket for the media channel

Perform CallAs the call progresses, RTCP, which runs alongside RTP (usually

on separate, consecutive UDP ports that are selected during call setup), can keep tabs on the media channel

ReleaseWhen the call concludes, H.225 enters its release state, signaling

an end to the media channel, an end to the H.245 application capabilities session, and an end to the call-accounting transaction on the gatekeeper

Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP) is an application-layer control protocol used to create, modify and terminate a communication session

Sessions can include audio, video and data streams

SIP Overview

The two components in a SIP system are user agents and network servers. Calling and called parties are identified by SIP addresses

User AgentsA SIP user agent is a client-end application continuing a

User-Agent Client (UAC) and a User-Agent Server (UAS.) These are known as a SIP client and a SIP server. The client initiates SIP requests as a user's agent. A server gets requests. A SIP server acts as a user's agent

SIP OverviewNetwork ServersTwo types of SIP servers proxy servers and redirect servers.

Proxy ServersAct on behalf of other clients and contains both client and

server functions. a proxy server interprets and can rewrite request headers before passing them on other servers. Rewriting ensures that the replies follow the same path back to proxy instead of the client

Redirect ServersAccepts SIP requests and sends a redirect response back to

the client containing the address of the next server

SIP OverviewAddressingSIP Uniform Resource Locators (URLs) provide addressing

similar to e-mail addressing. A SIP URL can have various forms and can include a telephone number, for example:

sip:someone@somewhere.com

sip:1-415-555-1212@somewhere.com

SIP Overview

SIP Methods and ResponsesINVITE: Start sessions and advertise endpoint capabilitiesACK: Acknowledge to the called SIP peer that an INVITE has

succeededBYE: This method is used when the call is completedCANCEL: This method is used during attempts to override a

prior request that has not yet been completedOPTIONS: Query a SIP peer for its capabilities information,

without actually establishing a media channelREGISTER: This method notifies the SIP server at which

endpoint a particular user can be reached

SIP Overview

SIP ResponsesInformational100 trying180 Ringing181 Call is being forwarded182 Queued

Success200 Ok300 Multiple choices

Client error400 Bad request401 Unathorized403 Forbidden408 Request timeout

482 Loop detected486 Busy here

Server error500 Server internal error502 Bad gateway

Global failure600 Busy everywhere603 Decline

SIP headerHeaders are used to transport the information to the SIP

entities. The main fields are:

- Via: shows the transport protocol used and the request route, each proxy adds a line to this field- From: shows the address of the caller.- To: show the called user address of the request.- Call-Id: Unique identifier for each call and contains the host address. It must be the same for all the messages within a transaction. - Cseq: begins with a random number and it identifies in a sequential way each message.- Contact : shows one (or more) address than can be used to contact the user- User Agent: The client agent who deals the communication.

SIP headerMessage Header

Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bK646464100000007343c52679000020a600000e45Content-Length: 0Call-ID: 911D32E5-EEDF-4572-B0B2-61B294636E88@192.168.0.100CSeq: 1 ACKFrom: "Prueba"<sip:20000@miasterisk.com>;tag=8922404614682Max-Forwards: 70Route: <sip:20001@192.168.0.1>To: <sip:20001@miasterisk.com>;tag=as0a27b928User-Agent: SJphone/1.60.289a (SJ Labs)Contact: <sip:20100@192.168.0.100:5060>;expires=3600

Inter-Asterisk Exchange (IAX) Protocol

The Inter-Asterisk Exchange (IAX) Protocol is a signaling protocol for VoIP networks, just like SIP and H.323. It also provides endpoint and trunk signaling

IAX is also NAT-proof, so dozens or hundreds of simultaneous calls from behind a masquerading firewall will function correctly, just like HTTP.

IAX is much more compact because it has been developed only for telephony applications

While a complete cycle of registration, call signaling, voice transmission, and tear-down can use several TCP and UDP ports and connections with SIP or H.323, IAX handles all of these functions using a single UDP port. When the IAX client (endpoint) registers with the IAX server or proxy, this UDP port is utilized. This same port is also utilized to place a call

Inter-Asterisk Exchange (IAX) Protocol  

The way IAX distinguishes between registration, signaling, and voice packets is by including headers and meta data in each packet