[Asterisk-Users] Cisco/Asterisk codec negotiation problems

[Alistair Cunningham]

As a followup for any who has the same problem, and searches the 
archives (don't you love finding the problem you have in the archive, 
but no-one followed it up?), check the following references:

http://lists.digium.com/pipermail/asterisk-dev/2005-April/011291.html

and the status of the updated code:

http://bugs.digium.com/bug_view_page.php?bug_id=0003346

Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/


Alistair Cunningham wrote:
> On more testing, I conclude that Asterisk isn't being very clever about 
> codec negotiation.
> 
> 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.
> 
> 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.
> 
> Is my understanding correct? Is this a weakness in Asterisk? Am I 
> missing something elementary?
>