[Asterisk-Dev] Re: Cisco/Asterisk codec negotiation problems

[Kevin P. Fleming]

Alistair Cunningham wrote:

> My understanding (possibly faulty) from experiments is this. If I have:
> 
> UA1 --> Asterisk --> UA2
> 
> and have disallow/allow entries in UA1's stanza in sip.conf, it seems
> that the first entry in the allow list is all that's used to choose the
> codec from UA1. Entries in UA2's stanza and SIP responses from UA2 are
> not used. If it turns out that UA2 can't support the codec that Asterisk
> chose for UA1, Asterisk attempts a translation. This occurs even if UA1
> and UA2 have a supported codec in common which isn't the one Asterisk 
> chose.

That is correct in the current code base, yes. I have been working on 
enhancements to make this more flexible, but the process of getting 
ready to move across country has hampered my coding time :-)

> If my understanding is correct, this is very inefficient. Worse, if one
> of the codecs is one it doesn't understand, such as G.729 (without
> chan_g729a.so) or G.723.1, Asterisk drops the call, even though it could
> have done pass through.

Right.