[SR-Users] Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP

Olli Heiskanen ohjelmistoarkkitehti at gmail.com
Thu Apr 3 21:32:57 CEST 2014


Hello,

Thanks, I'll give that a try and post back. I guess I install and run it
just like mediaproxy-ng?

I'll also try different sip clients like zoiper etc.

One thing that occurred to me based on the fact that the sdp is faulty, as
I did this test from the slides here:
http://www.slideshare.net/crocodilertc/webrtc-websockets
I wonder if the configuration (from page 53 in the slides) need
tweaking/fixing? Here's a snippet from my config:

failure_route[UA_FAILURE] {
if ( t_check_status("488") && sdp_content() ) {
if ( sdp_get_line_startswith("$avp(mline)", "m=") ) {
if ($avp(mline) =~ "SAVPF") {
$avp(rtpproxy_offer_flags) = "froc-sp";
$avp(rtpproxy_answer_flags) = "froc+SP";
} else {
$avp(rtpproxy_offer_flags) = "froc+SP";
$avp(rtpproxy_answer_flags) = "froc-sp";
}
#http://www.slideshare.net/crocodilertc/webrtc-websockets p.53/60
}
append_branch();
rtpproxy_offer($avp(rtpproxy_offer_flags));
t_on_reply("RTPPROXY_REPLY");
route(RELAY);
}
}
onreply_route[RTPPROXY_REPLY] {
        if (status =~ "18[03]") {
                # mediaproxy-ng only supports SRTP/SDES
                # early media won't work so strip it out now to avoid
problems
                change_reply_status(180, "Ringing");
                remove_body();
        } else if (status =~ "2[0-9][0-9]" && sdp_content()) {
                rtpproxy_answer($avp(rtpproxy_answer_flags));
        }
}



cheers,
Olli


2014-04-03 17:00 GMT+03:00 Richard Fuchs <rfuchs at sipwise.com>:

> Hi,
>
> This seems to be caused by an additional media stream (second m= line)
> appearing in the answer SDP, which is invalid according to RFC 3264.
>
> I'd like to invite you to try the upcoming new version of mediaproxy-ng
> instead, which has been renamed to rtpengine:
> https://github.com/sipwise/rtpengine
>
> It's still being worked on (including finalizing the name change), but
> it should handle those multi-stream cases much better (especially when
> WebRTC clients are involved), even though I can't guarantee that it will
> fix your problem in particular, as it's an RTC violation.
>
> cheers
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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>
>
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