[Asterisk-Users] P2P RTP without SIP re-invites

[Olle E. Johansson]

Let's go through how SIP works in Asterisk compared with a SIP Proxy. Remember that
Asterisk is not designed to be a SIP Proxy, it's designed to be a Multi-VOIP and PSTN
PBX, a quite complicated task.

(I'm not going into all details (ACK, TRYING, RINGING etc))

We have two SIP users, Alice and Bob.

Alice calls BOB, both connected to Asterisk:

* Alice's UA sends an INVITE to bob at Asterisk
* Asterisk checks if bob is a valid user reachable within the context
   allowed by Alice's account
* Asterisk answers the SIP call from Alice
* Asterisk initiates another SIP call to Bob's UA with a NEW Invite
* When Bob answers, Asterisk bridges the streams, performing codec conversion if necessary

In this scenario, we now have two different SIP dialogues (two separate SIP calls)

If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes
* Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it
   goes directly from Alice to Bob
Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so
the RTP media stream stays with Asterisk.

The benefit of this is that Asterisk acting as a user agent server (Alice) and client (bob)
can send early media to Alice, connect to voicemail or another extension than Bob if Bob had
issued a forward - maybe a H.323 connection or PSTN connection somewhere.

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With a SIP proxy we have the following scenario:

* Alice's UA sends an INVITE to bob at domain
* The proxy responsible for thte domain receives this and looks up bob in
   a user location or alias table
* The proxy *FORWARDS* the same INVITE to bob at currentlocation, maybe several different
   locations
* When Bob answers somewhere, the proxy cancels the call to the non-answering locations
   and forwards the OK to Alice
* Alice ACKs the OK to bob and the call is UP

In this scenario, there's only one SIP dialogue, between Alice and Bob with the
proxy in the middle of signalling, but acting as a proxy and not as a user agent
(the proxy can't and should not answer or originate calls).

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So, back to the original question, in a large installation (many users) - how do you off-load
Asterisk? There's no single truth here, but here's my opinion:

* If you are all on the same internal network, make sure the SIP phones
   support re-invites and use that.
* If you have users all over the Internet, use a SIP proxy as a front-end to Asterisk
   You will still be forced to handle a lot of RTP streams (because of NAT), but can
   distribute that over a SIP-proxy network with SRV records, DNS round-robin techniques
   or forcing the users to register with different proxies.

There's been a couple of suggestions that we should make Asterisk a good SIP proxy. If you
spend some time learning to understand Asterisk's architecture, you'll also understand
that this would not really work. I'm not saying the SIP channel can't be improved, I'm
just saying that it has to work with the rest of Asterisk's architecture.
I might be totally wrong, but my gut feeling is that Asterisk in combination with a
separate SIP proxy is a very powerful solution.

Clustering Asterisk servers somehow is also a good approach, but not here yet for SIP.

/O