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?
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