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

[Johansson Olle E]

16 okt 2006 kl. 20.32 skrev Brian Candler:

> On Mon, Oct 16, 2006 at 04:36:04PM +0200, Johansson Olle E wrote:
>>> 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.
>
> There is no re-INVITE; this is the very first (and only) INVITE  
> which sets
> up the call.
I know - but the same rules apply!

>
> I am talking about SVN, which behaves differently from 1.2 in this  
> regard.
>
> Please have a look at the SIP exchange which is attached as
> sip-codec-mismatch.txt to http://bugs.digium.com/view.php?id=8152
> and see if that makes the problem clearer.
>
As noted in the bug tracker - we need a SIP debug from Asterisk :-)

/O