← Back to team overview

rohc team mailing list archive

Re: [Question #403402]: Compress VoIP with ROHC

 

Question #403402 on rohc changed:
https://answers.launchpad.net/rohc/+question/403402

Didier Barvaux posted a new comment:
Hello,

> I study the data structure of the asterisk server,for I am going to know which
> lines to modify on chan_sip.c
> There are two methods on there, sipsock_read() and __sip_xmit(), which
> methods i want to add ROHC comressor in __sip_xmit and decompressor in
> sipsock_read().

I'm afraid that you are confused by SIP and RTP.

SIP (Session Initialization Protocol) is a protocol for negotiating a
media session (such as a VoIP call). A media session may be composed of
one or more several media streams, each one with different parameters
(audio/video codec, RTP parameters, UDP ports, and so on). The SIP
packets allow software to agree on codecs, bitrate...

One RTP (Real-Time Protocol) stream is for transmitting audio/video
codec with some additional sequence and timing information. RTP allows
software to re-order packets if they were not transmitted in order. RTP
also allows software to determine the correct timing for playing
audio/video in case network introduced some jitter (variation of
transmission delay).

See https://en.wikipedia.org/wiki/Session_Initiation_Protocol for SIP.
See https://en.wikipedia.org/wiki/Real-time_Transport_Protocol for RTP.


For a media session, there very few SIP packets and many more RTP packets. So, at the beginning, forget to compress SIP packets. Just compress RTP packets. As said in my previous message, RTP streams are handled in chan_rtp.c and res/res_rtp_asterisk.c.

Regards,
Didier

-- 
You received this question notification because your team ROHC Team is
an answer contact for rohc.