[asterisk-dev] Regarding SIP direct media (new topic)

[Johansson Olle E]

16 okt 2006 kl. 22.18 skrev Kevin P. Fleming:

> ----- Johansson Olle E <olle at voop.com> wrote:
>> As noted in the bug tracker - we need a SIP debug from Asterisk :-)
>
> I think it's pretty clear that this is a bug, of sorts. We are  
> offering codecs to the second endpoint based on our allow/disallow  
> settings for that endpoint, but with the media sent directly to the  
> first endpoint. This is bad.

I thought the original implementation sent the offer based on the  
incoming offer, not the current settings. I may be
wrong, but that was for sure the intent.

>
> As much as I hate to say it, I believe the method mentioned in the  
> other thread is the solution... send the 'direct media INVITE' to  
> the second endpoint, and if it responds with a final response that  
> indicates a bad codec selection (can we rely on that?) re-send it  
> with the entire list of codecs and no direct media path.

I'm not sure that's a good idea. Considering the amount of time it  
may take on a bad network to fail the first
INVITE, it may be more optimal to set up the call as we do in 1.2 -  
first through Asterisk, then investigate while
the call is up whether we can optimize the call path by sending re- 
invites. The dial() timer is ticking...

/O