10 Jul
2020
10 Jul
'20
10:26 a.m.
On 10/07/2020 04.59, Benjamin Flügel | vio:networks wrote:
Hey guys,
I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS
using WebRTC.
So far it works great on Firefox but have a specific problem with chrome, when I want to
make call from the softphone to another extension.
After anwsering the call Chrome/the softphone sends a BYE immediately, because this line
"a=rtcp-mux" is missing in the OK.
The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible
for the pbx-features.
Those are my rtpengine Flags for the Invite:
rtpengine_manage: replace-origin replace-session-connection trust-address
via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP
And those are the flags for the response, in this case the OK:
rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer
rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal
direction=external loop-protect
It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the
response. Can you help me get it to work?
You don't need to provide some of these options in your answer (neither
rtcp-mux nor the direction nor the protocol - the direction should be
specified in the offer). You should also provide the same via-branch
option in your answer as you did in the offer, especially if this is a
branched offer. In particular if this is a branched offer and the
via-branches weren't given correctly, then that would explain the
missing rtcp-mux.
Cheers