[SR-Users] kamailio + asterisk + webrtc

arish haque arish.haq at gmail.com
Fri Feb 1 13:58:43 CET 2019


Hi Daniel,
Within a network above tech stack is working fine for VOIP calls.
But when i am trying from outside there is no sign of RTP.


Cmd for rtp -
/sbin/rtpengine -p /var/run/rtpengine.pid
--interface=192.168.1.249\!106.51.78.78 --listen-ng 127.0.0.1:60000 -m
50000 -M 55000 --log-level 6 --log-facility local1

106.51.78.78 - public ip
192.168.1.249 - priv ip (kamailio + RTPEngine)

Kamailio config when call is b/w WebRTC --> WebRTC
rtpengine_manage("trust-address replace-origin replace-session-connection
direction=internal direction=external");

Here is SDP, captured on the client sitting outside.
v=0
o=root 1133801452 1133801452 IN IP4 106.51.78.78
s=Asterisk PBX 16.0.0
c=IN IP4 106.51.78.78
t=0 0
m=audio 50196 UDP/TLS/RTP/SAVPF 0 8 111 9 126
a=maxptime:60
a=ice-ufrag:5e13a1292fc0b7163d49328b7516f763
a=ice-pwd:4fa5e003195492fa165020c81eff9346
a=candidate:Hc0a801ac 1 UDP 2130706431 192.168.1.172 46810 typ host
a=candidate:Hc0a801ac 2 UDP 2130706430 192.168.1.172 46811 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256
AA:5A:51:BD:4C:53:65:E4:2B:EC:EB:BF:A9:07:DD:60:E3:46:D8:26:6D:04:C8:21:8B:B9:81:37:3A:EB:55:C0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=sendrecv
a=rtcp:50196
a=rtcp-mux
a=ptime:20
a=candidate:T7Tqd3oKM4NvATH3 1 UDP 2097152255 106.51.78.78 50196 typ host

Looking forward for help from you guys.

Thanks & Regards,
Arish Haque
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