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

[Low, Adam]

I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast.

So with that assumption I imagine a platform that would not get involved with the actual encoding/decoding of the RTP stream ensuring that only the SIP client's on each end of the call deal with RTP encoding with their dedicated DSP hardware. There is an alternative in mind that maybe I could utilise some old Dialogic DSP cards that we have but I suspect trying to get these working with Asterisk would be a lot of programming work that I probably couldn't manage, maybe I'm wrong ?

The SIP RE-INVITE mechanism is useful but I find problems when SIP clients are NAT'd (specifically SIP breaks and calls are not torn down correctly) and of course you lose a lot of monitoring (CDR's, etc.)and management capabilities provided by Asterisk when it is in the SIP signalling path.

I vaguely remember previous discussions on this and even a patch but I am unable to find anything in the archives, does anybody have any info on that ?

The conclusion I have come to is that I would try and patch the Asterisk code. The idea being that when the RTP parameters are negotiated that Asterisk would pass through the source address/port from each SIP client causing them to talk RTP directly. I intend to begin work on this this weekend but am I hoping that maybe somebody else has already achieved what I desire, anybody ?

Rgds,
Adam




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