Chan Kwang Mien wrote:
Thanks. By setting allow=g729, sip1 was able to connect to sip2.
Does that mean that Asterisk chooses the codec to be used between the
Caller and Callee ? In this case, since sip1 informs Asterisk that it
supports g.711 and g.729, Asterisk chooses g.729 since the Callee also
supports g.729. From the SIP Messages exchange, it doesn't seem that
Asterisk chooses the codec.
My previous setting was "allow=all". I was expecting "allow=all" to work
since that would also imply "allow=g729", isn't it ?
This really belongs on the -users list since it doesn't deal with
developing code. Moving it there now.
Each sip phone essentially negotiates a codec independently with
asterisk, and not as an end-to-end conversation.
When sip1 initiates a call, it exchanges sip packets with asterisk to
select a compatible codec. When asterisk places the call to sip2, it
exchanges sip packets with sip2 to select a compatible codec and it has
nothing to do with what sip1 negotiated.
You have two choices to correct the behavior. One, change the asterisk
definitions so as to show a preference (disallow=all, allow=g729,ulaw),
or, two, change the sip phone's definition to prefer g729 as its first
choice.
Even though many sip phones support multiple codecs, the negotiation is
very simple in that it offers up its first choice codec (only), and if
asterisk supports that first choice, that's what is used. Highly
dependent on the sip phone manufacturers coding.
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