Deployment could be quite simple,
Create a B2BUA or proxy that TSIPs and users can deploy to forward between legacy SIP UA's, until other SIP stacks can catch up.
Regards
Lafras
----- Original Message -----
From: Lafras Henning
To: pjsip at pjsip.org
Sent: Wednesday, October 03, 2007 8:31 PM
Subject: [pjsip] SIP trunking
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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