[asterisk-dev] Regarding SIP performance

[Johansson Olle E]

16 okt 2006 kl. 16.25 skrev Brian Candler:

> On Mon, Oct 16, 2006 at 03:34:10PM +0200, Johansson Olle E wrote:
>> Or, set up the call as normal - one call leg to Asterisk, transcode
>> and another
>> call leg and media stream to the other device.
>>
>> That should be the case here.
>
> But the point is - how do you know when you need to do that?
>
> At the moment (SVN), Asterisk is passing on an INVITE with SDP  
> pointing to
> the first endpoint. That works sometimes, but not other times,  
> depending on
> which codecs the second endpoint supports. But you have to know  
> whether to
> try this before you send on the INVITE.

The same rules apply as for the previous re-invites. So i guess we  
either have
a bug or you have a bad configuration that makes this happen.

/O