I’m install and config asterisk, webrtc in vmware. I’m login webrtc client with chrome and call to IVR. Asterisk always send rtp to external ip.
I do not hear sound from the browse. the system can be connected to stun server ?
P/S: asterisk and client on LAN
Thanks
RTP debug log:
– Executing [1234@outgoing:1] NoOp(“SIP/6001-00000000”, “en”) in new stack
– Executing [1234@outgoing:2] Playback(“SIP/6001-00000000”, “demo-congrats”) in new stack
Sent RTP packet to 113.190.243.97:11224 (type 00, seq 011325, ts 000160, len -000013)
– <SIP/6001-00000000> Playing ‘demo-congrats.gsm’ (language ‘en’)
Sent RTP packet to 113.190.243.97:11224 (type 00, seq 011326, ts 000320, len -000013)
Sent RTP packet to 113.190.243.97:11224 (type 00, seq 011327, ts 000480, len -000013)
Sent RTP packet to 113.190.243.97:11224 (type 00, seq 011328, ts 000640, len -000013)
[Oct 8 09:31:45] WARNING[1580][C-00000000]: app_playback.c:493 playback_exec: Playback failed on SIP/6001-00000000 for demo-congrats
– Executing [h@outgoing:1] Hangup(“SIP/6001-00000000”, “”) in new stack
== Spawn extension (outgoing, h, 1) exited non-zero on 'SIP/6001-00000000’
Help me please !!!
You have not stated what version of Asterisk or provided configuration either. Those would be needed by anyone wanting to look at this.
[quote=“haihai2212”]I’m install and config asterisk, webrtc in vmware. I’m login webrtc client with chrome and call to IVR. Asterisk always send rtp to external ip.
[/quote]
Configure correctly your Nat settings in asterisk and or check the stun server at your sipml5 client, this issue is to usual and has been reported a lot times in this forum and in the doubango group.
Thank for reply,
I’m using asterisk 13.3.2, sipml5, chrome 45.0.2454.101
help me please !!!
my system config:
asterisk:
sip.conf
[general]
realm=172.20.109.203
udpbindaddr=0.0.0.0:5060
transport=udp,ws,wss
[6001]
host=dynamic
secret=6001
context=outgoing
type=peer
encryption=yes
avpf=no
icesupport=no
transport=ws,wss,udp
directmedia=no
disallow=all
allow=all
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
qualify=yes
;extension for softphone application
[6002]
host=dynamic
secret=6002
context=outgoing
type=peer
transport=udp
directmedia=no
disallow=all
allow=all
rtp.conf
rtpstart=10000
rtpend=20000
httpd.conf
enabled=yes
bindaddr=0.0.0.0
Webphone cline:
Expert settings
WebSocket Server URL
ws://172.20.109.203:8088/ws
SIP outbound Proxy URL
udp://172.20.109.203:5060