[Asterisk-Dev] Input and testing.

[Kevin P. Fleming]

Brian West wrote:
> The SDP has nothing to do with this.  The SDP doesn't even report the rtp
> frame length.  RTP is just RTP no matter the size. Asterisk deals with
> inbound 100ms frames or any other variation I threw at it. I think the only
> exception to this is SPEEX which needs out of band means to tell the frame
> sizes that will be sent if you pack more than one in a packet.  ILBC has two
> restrictions on putting more than one frame per RTP packet which is you
> can't mix 20ms and 30ms frames in the same packets and you can't exceed the
> MTU per packet.

Yeah, I don't know what I was thinking of there... must have been the 
long weekend playing tricks with my memory.

However, given that you can only affect this on one end of the link, 
right (unless the SIP endpoint has a corresponding setting)?

> As far as exposing this to the channel driver I'm not even going to attempt
> it without someone with some say giving me an approved approach.  I'm not
> going to spend time trying to solve the same problem three different ways.
> I was thinking of adding a param to ast_rtp_write.

Wouldn't it make more sense to pass in the parameter when the RTP 
session is being 'created', rather than every time a packet is written?