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.323 SIP IAX

H.323 H.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 Protocols FeatureProtocol Call SignallingH.225 Media ControlH.245 Audio CodecsG.711, G.722, G.723, G.728, G.729 Video CodecsH.261, H.263 Data SharingT.120 Media TransportRTP/RTCP

H.323 Elements H.323 elements include terminals, gateways, gatekeepers and Multipoint Control Units (MCUs). Terminals Also known as endpoints, terminals provide point-to-point and multipoint conferencing for audio, video and data Gateways 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 Gatekeeper Gatekeepers 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/Teardown To 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 entity Proceeding: 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 Process Alerting: A backward message sent from the called entity to inform that called party ringing was initiated Connect: 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 Negotiation After 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 Process Open Media Channel Once capabilities negotiation has succeeded, RTP Control Protocol (RTCP) establishes a UDP socket for the media channel Perform Call As 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 Release When 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 Agents A 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 Overview Network Servers Two types of SIP servers proxy servers and redirect servers. Proxy Servers Act 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 Servers Accepts SIP requests and sends a redirect response back to the client containing the address of the next server

SIP Overview Addressing SIP 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 Overview SIP Methods and Responses INVITE: Start sessions and advertise endpoint capabilities ACK: Acknowledge to the called SIP peer that an INVITE has succeeded BYE: This method is used when the call is completed CANCEL: This method is used during attempts to override a prior request that has not yet been completed OPTIONS: Query a SIP peer for its capabilities information, without actually establishing a media channel REGISTER: This method notifies the SIP server at which endpoint a particular user can be reached

SIP Overview SIP Responses Informational 100trying 180 Ringing 181 Call is being forwarded 182 Queued Success 200Ok 300Multiple choices Client error 400Bad request 401Unathorized 403Forbidden 408Request timeout 482Loop detected 486Busy here Server error 500Server internal error 502Bad gateway Global failure 600Busy everywhere 603Decline

SIP header Headers 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 header Message Header Via: SIP/2.0/UDP :5060;rport;branch=z9hG4bK c a600000e45 Content-Length: 0 Call-ID: CSeq: 1 ACK From: "Prueba" ;tag= Max-Forwards: 70 Route: To: ;tag=as0a27b928 User-Agent: SJphone/ a (SJ Labs) Contact: ;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