[asterisk-dev] SIP: handling multiple m=video or m= audio lines

[Olle E. Johansson]

11 sep 2009 kl. 13.27 skrev Vadim Lebedev:

> Olle E. Johansson wrote:
>>
>> In short, just forwarding FMTPs sound easy but is not. I don't think
>> it's part of this short term project focused on fixing the SDP stream
>> handling, but part of the new media architecture/codec negotiation
>> project that we all wish we had proper  resources to be able to  
>> start.
>>
> While i agree that simple forwarding of fmtp is  not a long term
> solution and is really a hack,
> I've hunch that it will be sufficient for 70% of SIP video users.....

Yes but Kevin and I will be part of the team that will have to handle  
all the bug reports for the 30%. And manage the code. And be a bit  
ashamed.

It has gotten very hard in a large project that Asterisk has become to  
implement things half-heartedly. It was something we could do in the  
early days because users would accept us saying "no, we never  
understood how that should work" or "that's stupid so we ignored it".  
Asterisk has grown up and people expect that if we add a feature, we  
give it our best and implement it properly.

Success has raised our bars, for good and bad.

I suggest that we try to take this step by step. Just for  
experimentation, go ahead and come up with a patch so that it can be  
made available and give us something to discuss. At least, the 70%  
will be happy that they can patch their Asterisk with something less  
intrusive than videocaps, that has solved the very same problem in  
Asterisk years ago ;-)

/O