[pjsip] SIP trunking

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Hi Benny,
A general VOIP question,

I see in the internet a lot of reference to SIP trunking, but this refers to
just using normal SIP to communicate a number of channels to a TISP.

IAX/2 fills a niche mainly to optimise bandwidth by sharing IP overhead.
Needed when in high compression codecs, overhead is 50% of the traffic.

Surely we don't need a totally different protocol to do this?
It would be better to concatenate SIPand RTP packets (going to a single destination/proxy)
into a single UDP/TCP/TLS payload with change to the transport layer of the SIP stack (and publish a RFC on it).

- The main objective is to share headers.
- Optionally some SIP content compression could be added.
- I don't think RTP content compression would have much benefit(can't beat speex :).
- Implemented in the transports layer.

Or do you think the gains would be inconsequential?

Or do you think such is better left for external solutions such as VPN -
  although I don't think that would allow for optimal bandwidth solution.

Or is it a stupid idea?

If not let's call this SIPTP - SIP Trunk Protocol - logical hey?

Not to be confused with 
trans-di-?-hydridobis(silyl)bis(trialkylphosphine)di-platinum complexes (SiPtP)2 -
and we thought programmers were geeks! ;)

Regards
Lafras


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