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

[Alistair Cunningham]

Kevin,

Thank you for confirming what I suspected. Do you have any recommended 
workarounds? Do you have an expected timeframe for an official release 
with your enhanced code?

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


Kevin P. Fleming wrote:
> 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.
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