RTP packet flow

Does RTP packets flow between endpoints once the sip call is setup in an Ip Pbx. and hosted Ip Pbx.

James

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v
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If the PBX and endpoints are setup correctly the RTP data stream is redirected to go between the endpoints. You really don't want the sip/rtp proxy to stay in the data path. It just chews up bandwidth on the server and adds delay and jitter. For more information google for "sip reinvite" without the quotes.

-wolfgang

Reply to
Wolfgang S. Rupprecht

"Wolfgang S. Rupprecht" wrote in message news: snipped-for-privacy@bonnet.wsrcc.com...

...and the endpoints are not both behind NAT, which unfortunately happens very often...

Well, there are cases when have to. If DTMF signalling is passed over RTP, as per RFC2833, the removal of the PBX from the data path makes it unable to respond to requats such as "call transfer"... Same story if the PBX performs other useful functions like transcoding between endpoints that do not share a common codec. You may argue that this is not a case of endpoints correctly set up, but it's a reality especially in a peer-to-peer, providerless environment without a guaranteed interoperability baseline.

Cheers --

Enzo

Reply to
Enzo Michelangeli

True. Putting SIP phones behind NAT isn't set up correctly. They really should have a real IP.

(My feeling is that NAT is evil and breaks way to many symmetrical protocols.)

True. This is exactly why one wants to route the DTMF via SIP-Info messages. One needs to keep the SIP proxy "in the loop".

I've been running with reinvites active for 2 years now. It can be made to work well and does in fact help quite a bit with jitter/drop-outs when running a soft-pbx on a general purpose computer. Having reinvites working is very important when running off-site extensions. It would be very wasteful to haul the RTP/ulaw stream onto one's net and then send it right out the same connection. RTP isn't that efficient and a 64kbit/sec ulaw stream ends up taking

96kbits/sec on the wire (if I'm remembering the numbers correctly). That comes out to 192kbits/sec just for looping the voice stream through the proxy.

As an aside, the first issue with each phone needing its own IP (or suffering higher delays and jitter) is what I think will drive ipv6 adoption. ISP's, at leas in the US, are in general very stingy with how many IP's they hand out to customers. With ipv6 it is hard for them to not hand out a full set of 2^64 IP's to each customer.

-wolfgang

Reply to
Wolfgang S. Rupprecht

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