[OpenSIPS-Users] [NEW] SDP codec manipulation feature

[Olle E. Johansson]

28 jul 2009 kl. 16.51 skrev Alex Balashov:

> Olle E. Johansson wrote:
>
>> As far as I know, there's no way in SIP you can determine what  
>> codec  actually was used if the offer/answer resultet in multiple  
>> codecs.
>
> I was just going to say that.  Even if you mimic the exact algorithm  
> used by the offer and answer side, since there is no knowledge of  
> their intrinsic codec capability set, there's no way to know what  
> the decision rendered ultimately is.
>
>> Also note that during a call, the codec may change.
>
> By means other than re-INVITEs? (Which can also be inspected for SDP.)

Well, if you end up with an offer/answer with G711/GSM/G722 any  
endpoint can choose freely from that list, in the order provided. You  
might very well have a call with different codecs in each direction.  
Check with wireshark - this happens. A b2bua like asterisk should try  
to either follow the list or try to optimize it's own processing and  
avoid transcoding of the audio path, which might result in interesting  
scenarios, depending on configuration or actual code decisions. On  
large scale installations, the service provider wants to optimize the  
b2bua processing, ie cpu usage by avoiding transcoding at the cost of  
the endpoints... :-)

/O