N/ACIP: Simplifying Codec Connections

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Can your box play nice with dozens of other codecs?

Another aspect of the operation of a N/ACIP-compliant codec is the use of SIP to establish a session between two endpoint devices (i.e., two audio-over-IP codecs).


What is SIP? Let's take a look at SIP and see why its use makes sense, along with some particulars about how it works.

SIP stands for Session Initiation Protocol. It's an application layer protocol that is agnostic about what transport layer it runs on top of — for example it will work with either UDP or TCP. The job of SIP is the creation, modification (if needed) and termination of sessions between two endpoints — which would be two IP codecs. By the way, in the parlance of SIP, these endpoints are also known as UAs (user agents).

Consider this common scenario, which should clarify the advantage to using SIP.

Say you are a news reporter, and your typical means of communicating with your HQ is by way of an IP codec that can work over wired networks or Wi-fi networks. As you move around and connect to various networks, your IP codec will necessarily have to get a new IP address every time (likely by way of DHCP). For HQ to initiate a connection with you, it would have to have an updated IP address each and every time you moved.

This would be like using a cell phone that had its number change every time you got into a different cell. Not very practical — you'd be tough, if not impossible, to reach.

With SIP, your codec gets a URI (Uniform Resource Identifier). This looks and sounds very much like a Web address, probably because SIP is modeled on HTTP. The format of the URI is SIP:username@domain.com. For our example then, say our news reporter has a URI of SIP:newsguy@bignewsnetwork.com. Every time newsguy gets on a different network, his codec registers its new IP with a registrar server that functions very much like a DNS server. (HQ could very well be hosting its own SIP server, which, among other things, performs the registrar server duties. This server would live on the public side of a firewall on the network.) When HQ wishes to contact newsguy, it does so by dialing his URI on the codec at HQ. SIP uses the updated information in the registrar server to come up with the actual IP.

Aside from putting the two endpoints in contact with one another, SIP also sees to it that they agree on the codec to be used, thus assuring that they actually do pass audio from one end to the other. After the session is established, SIP goes idle; the two endpoints connect directly with one another for the actual passage of the payload data. Once the session is over, SIP is also used to tear down the link and effectively end the communications.

-- continued on page 3

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