[asterisk-dev] Why does chan_sip disable early bridging???

[Sergio Garcia]

Thanks for the clarification. 
Probably I'm again wrong, but I assume that the main problem is knowing
if the remote peer has compatible codecs in order to keep asterisk in the
media loop or not. Could it be solved if we implement the option of
sending INVITEs without SDP and sending it in the ACK?

BR
Sergio

---------- Original Message ----------------------------------
From: Olle E Johansson <olle at voop.com>
Date:  Wed, 3 Oct 2007 10:16:46 +0200

>The early bridge is an asterisk concept where we set up the call  
>directly
>between two endpoints instead of making the decision later and sending
>re-invites.
>
>We do support early media but not with PRACK. That's a different thing.
>
>/O
>3 okt 2007 kl. 09.57 skrev Sergio Garcia:
>
>>
>>
>>
>> ---------- Original Message ----------------------------------
>> From: Olle E Johansson <olle at voop.com>
>> Reply-To: Asterisk Developers Mailing List <asterisk- 
>> dev at lists.digium.com>
>> Date:  Wed, 3 Oct 2007 09:44:11 +0200
>>
>>> The reason early bridge was disabled by default (but can be enabled
>>> in the configuration)
>>> was that it had some pretty large design flaws when used with the SIP
>>> offer/answer model.
>>> There's no way to signal back what the answering device offers in the
>>> SDP and in those
>>> cases the call setup might fail miserably. If you test it and it
>>> works, keep it enabled in
>>> the configuration.
>>>
>>> The videocaps branch has some ideas on how to signal back to the  
>>> caller
>>> properly and this was discussed on the developer meeting in  
>>> Atlanta, but
>>> seems to have gotten stuck somewhere in the process.
>>>
>>> /O
>>>
>>
>> Hi Olle, I'm just curious about the issue..
>>
>> Could you explain a bit better what are those designs flaws?
>> The early media in SIP is covered by the PRACK method, is there any  
>> issue
>> that prevent us of implement it?
>>
>>
>> Best regards
>> Sergio
>>
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>
>