1 Nov
2018
1 Nov
'18
6:40 p.m.
Dear Kamailians,
I just opened a merge request for a new module called RTP Media Server
I understand that integrating an RTP/Media endpoint directly in Kamailio
may seems strange at first.
Let me share my thoughts ...
By combining Kamailio, oRTP and mediastreamer2
this module is providing some very good foundation to support RTP endpoint
and various media processing features.
This seems like a great idea for various use cases like IoT, IVR or other
specific needs.
Kamailio is still handling everything related to SIP/SDP (the module is
adding some SDP parsing, they could be integrated in the core if this seems
fit).
By doing integration with a library we can benefit from the scripting
engine and probably KEMI.
This does not means that everyone should run this module on a Kamailio
sever acting as a load balancer, of course CPU intensive operations like
resampling and encoding may be taking place on the server.
oRTP is providing RTP endpoints compliant with (RFC 3550)
MediaStreamer2, even if written in C is quite a high level library, because
it is implementing a framework for audio processing using graphs of
filters, filters can be to do various things.
Similar to playing with lego blocks :)
- Support for most free and some non free codecs can be added easily.
- Work to bridge calls is already in progress in the module.
- Mobile phone support ARM CPU and other embedded scenario should be
supported
Mediastream2 is creating one thread per call "msticker", this can work
smoothly in Kamailio even if it is forking processes.
Shared memory allocation is supported using wrapper around malloc/free used
by the libraries.
This project was started last year, I exposed the idea at
KamailioWorld2017, the feedbacks I god where not all positives :)
But we did speak about it again at ClueCon2018 and I was encouraged to
complete the work and propose a MR.
The module is not working and was tested with several softphones.
I think it is time to submit it ... before considering adding more features
or investing more time in it.
Julien
Example config using the features already implemented
event_route[rms:start] {
xnotice("[rms:start] play ...\n");
rms_play("/tmp/reference_8000.wav", "rms:after_play");
};
event_route[rms:after_play] {
xnotice("[rms:after_play] play done...\n");
rms_hangup();
};
route {
if (t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
if (is_method("INVITE") && !has_totag()) {
if (!rms_answer()) {
t_reply("503", "server error");
}
}
if (is_method("BYE")){
rms_media_stop();
}
}