[asterisk-dev] [RFC] optimisinf RTP traffic?

[John Martin]

> 
> ssize_t sendfile(int out_fd, int in_fd, off_t *offset, size_t count);
> 
> sendfile() can do zero copy transmission of data between sockets,
> meaning no in-memory copying of the data, and having the NIC (if
> supported) generate checksums as well. This will increase the number
> of bridgable calls by, well, although I obviously haven't tested it,
> quite a lot.
> but then, since sendfile() just sends the data, this means you won't
> be able to Monitor() the calls, you won't be able to touch the RTP
> stream. someone said 'you need to update RTP timestamps', but I
> disagree, since dejittering is done in the endpoint, and then it's
> the client's timestamps that are important. My setup is client -
> asterisk - asterisk - pstn, and it's the pstn gateway dejittering
> (using 1.2.9.1 with the jb patch from asterisk-backports.org).
> 

It's possible that directly copying RTP from one port to another could
work in some situations, but some paths through Asterisk (for instance
coming off a queue with moh, or voicemail transfer to operator etc)
would cause changes in the ssrc and sequence number jumps. Some
endpoints don't like either or both.

John