[asterisk-dev] Regarding SIP performance

[Johansson Olle E]

16 okt 2006 kl. 12.50 skrev Brian Candler:

> On Mon, Oct 16, 2006 at 10:29:52AM +0100, Brian Candler wrote:
>> I note Asterisk offers extra codecs in the second leg, because it  
>> could
>> transcode if necessary; presumably if the second handset chose a  
>> codec not
>> supported by the first, Asterisk would need to re-INVITE to set up  
>> a two leg
>> call. I've not tested this though.
>
> I've tested it now, and in my SVN build, which is about 10 days out  
> of date,
> it appears to be broken.
>
> Full description posted at http://bugs.digium.com/view.php?id=8152
>
> In summary: the two endpoints think they can both use different  
> codecs at
> the same time, with the RTP stream running directly between them :-(

If they're not compatible, we should not re-invite or set up the RTP  
directly.
I'll check the bug report.

/O


---
Olle E. Johansson * Asterisk Evangelist, developer * VOOP A/S
olle at voop.com