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Between dial tone, number lookup, ringing, and busy signals, there’s quite a lot that has to happen before you even start speaking and what most people think of as a “phone call” even occurs. Call setup protocols not only do these things, but also perform after-call resource cleanup and statistical reporting.
Each protocol uses TCP or UDP, and a well-known port or ports for communication. Some call setup protocols are used primarily for communication between endpoints (IP phones) and call servers, while others allow for communication between call servers and voice gateways handling calls to and from the PSTN.
These messages, which vary in size and number, handle functions like the mapping of phone numbers to IP addresses, generating dial tones and busy signals, ringing the called party, and hanging up.
Many different call setup protocols, some standardized, and some proprietary are in widespread use in VoIP deployments. We’ll describe some of the most popular call setup protocols below:
H.323
The call setup protocol H.323 is standardized by the International Telecommunications Union (ITU). It is a family of telephony-based standards that can be used by endpoints for both voice and video conferencing.
H.225 and H.245 are two of the protocols in the H.323 family that are involved in call setup. H.225 uses TCP and port 1720 for communications. H.245 uses TCP as well, but the ports used are dynamically negotiated during the H.225 phase of the setup.
H.323 has been widely deployed and, was among the first call setup protocols used for VoIP calls. In a VoIP environment, H.323 is commonly used by voice gateways to connect the IP network to the PSTN.
However, setting up a call with H.323 can require many TCP flows and handshakes. Some implementations of the protocol provide a “fast start” capability to bypass some of the normal handshaking in an effort to speed up call setup performance. Be aware of this fact if you are investigating network performance issues during call setup.
Some additional configuration on the voice gateway is required for H.323 because the gateway maintains the information about how calls are routed. H.323 is a peer-to-peer protocol that lacks centralized configuration.
MGCP
The Media Gateway Control Protocol (MGCP) is another popular call setup protocol, standandardized by the IETF in RFC 3435. In MGCP, the endpoints typically don’t use the MGCP protocol to control the phone call – it’s most often used to allow a call server to control a voice gateway connection to the PSTN over UDP port 2427. Because the call server is controlling the gateway, the call control intelligence resides in the server – and with MGCP, the call routing information is also configured in the server.
Q.931
While Q.931 wasn’t designed specifically for call setup, it supports call setup by defining Layer 3 signaling information for ISDN Primary Rate Interfaces (PRI), which are commonly installed in voice gateways to provide connectivity to the PSTN. In an ISDN PRI, a separate TCP session on port 2428, often referred to as a “PRI Backhaul,” is used to communicate additional information between the voice gateway and call server. This provides the ISDN channel signaling information to the call server where the gateway is registered.
SIP
SIP (Session Initiation Protocol) is a call setup protocol developed by the IETF in RFC 3261 (and in many other RFCs too numerous to mention). SIP represents typical data-networking protocols: lightweight, relatively easy to understand and implement, ASCII-based. Vendors such as Cisco, Nortel, Avaya and Microsoft are all using some form of SIP.
Although SIP can be carried over TCP or UDP, most implementations use TCP port 5060. Secure SIP uses TCP and port 5061. SIP messages are text-based and generally follow a request-response structure – much like HTTP.
Another interesting aspect of SIP is its ability to connect IP networks to the PSTN or other IP networks. Carriers are offering SIP trunking packages to enable an enterprise to connect by means of a session border controller (SBC) device to the PSTN or to another IP network.
Because SIP can not only be used for voice call setup but also for video and instant messaging setup, our view is that SIP is a key enabler of unified communications.
In addition to the standardized call setup protocols discussed above, vendors have developed their own proprietary protocols.
Cisco Skinny Client Control Protocol (SCCP)
Cisco phones typically use the SCCP (Skinny Client Control Protocol) for call setup. It’s a simple, lightweight call setup protocol for endpoints controlled by Cisco Unified Communications Manager). “Skinny” passes messages using TCP port 2000. It can also be secured using using Transport Layer Security (TLS) which uses TCP port 2443.
Nortel UNIStim
Nortel phones typically use the UNIStim protocol to communicate with call servers. UNIStim is a UDP-based request/ response protocol and has reliability built in at the application level. UNIStim uses port 5000 for communications.
Right now, there isn’t a single call setup protocol that dominates the market – and all the protocols here are used by VoIP equipment in varying degrees. However, the trend is moving slowly towards SIP as the standard protocol of choice.
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