[asterisk-users] Asterisk RTP bridging

[Alex Balashov]

On Mon, 13 Aug 2007, Kutman.DK at forces.gc.ca wrote:

> I have a small LAN network connected through an Asterisk Server 
> (Trixbox).  I was looking to create my own custom made softphones, and I 
> have been looking into how to transmit and receive via RTP.  Would 
> anyone know how the Asterisk RTP bridging works, and if there is any 
> documentation on it?  How is the RTP stream routed through the Asterisk 
> server?  Do I just give it the endpoints and then the audio call is 
> transmitted directly between the two machines?

   Pretty much.  All the RTP media endpoint stuff is negotiated via SDP
(Session Description Protocol), which rides in the payload of the INVITEs,
200 OKs and provisional 18x messages.  This determines the UDP ports for
RTP send/receive on both ends of the leg (they are one and the same port,
I believe), the packetisation duration (the audio interval to be buffered
into a single datagram in the appropriate codec), the codec itself, etc.

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671