[SR-Users] RTPEngine a=rtcp-mux

[Igor Olhovskiy]

Getting same result..

v=0
o=- 1334974643 4 IN IP4 X.X.X.X
c=IN IP4 X.X.X.X
t=0 0
m=audio 41572 UDP/TLS/RTP/SAVPF 0 9 126
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:41573
a=setup:active
a=fingerprint:sha-1 79:28:0E:B7:44:65:8F:69:9A:B5:A6:A4:FF:79:A8:78:51:X:X:X
a=ice-ufrag:skh8lj8m
a=ice-pwd:FE5tjyUuBmWDaVhE3zcsYgZr52
a=candidate:YLrFRpDB3kwIFygB 1 UDP 2130706431 120.76.27.231 41572 typ host
a=candidate:YLrFRpDB3kwIFygB 2 UDP 2130706430 120.76.27.231 41573 typ host
m=video 0 UDP/TLS/RTP/SAVPF 96 98 100 102 127 97 99 101 125
c=IN IP4 X.X.X.X


Regards, Igor

On 2 июня 2017 г., 14:57 +0300, Richard Fuchs <rfuchs at sipwise.com>, wrote:
> On 02/06/17 07:40 AM, Igor Olhovskiy wrote:
> > Hi!
> >
> > I’m trying to make RTPEngine to work with calls Browser <-> SIP.
> > Call From SIP to Browser working perfectly, but when calling to SIP from Browser, on 200 OK (sdp) I don’t receive
> > ...
> > a=rtcp-mux
> > ...
> > lines, but it was told to RTPEngine on reply_route to add this. And this line is mandatory for me, cause I don’t control JS softphone on other side
> >
> > Reply route in a case of Browser->SIP call looks like
> >
> > reply_route[REPLY_R] {
> > …
> >   rtpengine_answer("internal external UDP/TLS/RTP/SAVPF ICE=force rtcp-mux-offer");
> > ...
> > }
> >
> > Also I’ve tried with rtcp-mux-accept but still no luck
>
> You cannot "offer" rtcp-mux an in "answer".
>
> I believe what you want is to use rtcp-mux-demux in the offer.
>
> Cheers
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20170602/0e2fad3f/attachment.html>