[Asterisk-Dev] Payload size for native bridged conversations

[Alf Thomas Nilsen]

I'm trying to control the payload size of conversations. Default seems to be
20 milliseconds for G.729A, but for regular conversations I've been able to
change this to 40 milliseconds. This lags conversations somewhat but
bandwidth usage is around 10kbits lower. I do this by modifying the call to
ast_smoother_new() on line 1225 in rtp.c:

 

 case AST_FORMAT_G729A:

                if (!rtp->smoother) {

                                    /* rtp->smoother = ast_smoother_new(20)
*/

rtp->smoother = ast_smoother_new(40);

 

This has no effect on the traffic between two Asterisk-servers in a bridged
conversation. The following is a crude drawing of a G.729A SIP conversation.
Both Asterisk servers have G.729A codec installed so they're able to bridge
this codec and the client sends voice in G.729A:

 

 

                                  45 bytes packets

      Depending on client             35 kbit

      ------------------->        ----------------->

Client                    Asterisk                  Asterisk

      <-------------------        <-----------------

      ast_smoother_new(40)         45 bytes packets

        52 bytes packets              35 kbit

            19 kbit

 

Following is CLI printout from the Asterisk server which the client
registers with:

 

  == Spawn extension (context, 1000, 1) exited non-zero on 'SIP/client-e097'

    -- Executing Dial("SIP/client-6b02", "SIP/asterisk2/1000") in new stack

    -- Called asterisk2/1000

    -- SIP/asterisk2-559b answered SIP/client-6b02

    -- Attempting native bridge of SIP/client-6b02 and SIP/asterisk2-559b

 

What I want is to control the payload of the traffic between the Asterisk
servers. Reinvites is not allowed in this particular system.

 

Best regards,

Alf Thomas Nilsen

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