No Audio Webrtc call to Asterisk

Asterisk Asterisk General
1

Chrome Canary – Sipml5 – ws --> Kamailio - udp -> Asterisk 12.5 (AWS)

When I call to Asterisk, my call shows connected but I get no audio:

  1. I can’t see the public IP address ICE candidates in the 200 OK SDP from *.
  2. Try google stun and same issue.
  3. Error is generated during call initiation:

[2014-09-14 06:05:01] ERROR[23029]: pjsip:0 <?>: icess0x7fa8bc0 …Error sending STUN request: Invalid argument
[2014-09-14 06:05:17] NOTICE[23049]: chan_sip.c:28904 check_rtp_timeout: Disconnecting call ‘SIP/SipRegistrar-00000000’ for lack of RTP activity in 16 seconds

  1. SIP trunk is configured like this:

host=172.31.27.85
port=5060
type=peer
fromdomain=ramenlabs.io
avpf=yes
encryption=yes
icesupport=yes
nat=force_rpot,comedia
context=from-pstn
transport=udp,ws
directmedia=no
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlssetup=actpass

Stun shows right Public IP address

ip-172-31-24-110*CLI> stun show status
Hostname Port Period Retries Status ExternAddr ExternPort
stun.counterpath.net 3478 30 3 OK A.B.C.D 57177

SIP 200 Ok in my Chrome Browser,

recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=37111;received=172.31.22.2;branch=z9hG4bKDDDaByCi6JTZTmFrK5N1LJHsGBS9C7kJ
From: "Gonzalo Gasca Meza"sip:gogasca@ramenlabs.io;tag=EW3o6gulfrf4ad1Gc7PC
To: sip:1000@ramenlabs.io;tag=as5ae50e6a
Contact: sip:1000@A.B.C.D:5070
Call-ID: 52c9c82c-9079-54aa-3603-c1d1c7d02670
CSeq: 65151 INVITE
Content-Type: application/sdp
Content-Length: 839
Record-Route: sip:172.31.27.85;r2=on;lr=on;ftag=EW3o6gulfrf4ad1Gc7PC
Record-Route: sip:172.31.27.85:5062;transport=ws;r2=on;lr=on;ftag=EW3o6gulfrf4ad1Gc7PC
Server: Llamato Media Server
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

v=0
o=root 2031387416 2031387416 IN IP4 A.B.C.D
s=Asterisk PBX 12.5.0
c=IN IP4 A.B.C.D
t=0 0
m=audio 54716 RTP/SAVPF 0 8 111 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:60
a=ice-ufrag:180614046b03a5f23e427da233f50162
a=ice-pwd:15191276696dede74834d208375270b3
a=candidate:Hac1f186e 1 UDP 2130706431 172.31.24.110 54716 typ host
a=candidate:Hac1f186e 2 UDP 2130706430 172.31.24.110 54717 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 BA:E8:BE:66:C6:36:8D:C5:0C:67:A3:CE:15:69:5F:96:DB:E1:B2:E2:B0:F4:59:3E:1E:C9:8F:54:BB:7C:DF:20
a=sendrecv

SIPml-api.js?svn=224:1 State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE
SIPml-api.js?svn=224:1 setRemoteDescription(answer)
v=0
o=root 2031387416 2031387416 IN IP4 A.B.C.D
s=Asterisk PBX 12.5.0
c=IN IP4 A.B.C.D
t=0 0
m=audio 54716 RTP/SAVPF 0 8 111 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:60
a=ice-ufrag:180614046b03a5f23e427da233f50162
a=ice-pwd:15191276696dede74834d208375270b3
a=candidate:Hac1f186e 1 UDP 2130706431 172.31.24.110 54716 typ host
a=candidate:Hac1f186e 2 UDP 2130706430 172.31.24.110 54717 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 BA:E8:BE:66:C6:36:8D:C5:0C:67:A3:CE:15:69:5F:96:DB:E1:B2:E2:B0:F4:59:3E:1E:C9:8F:54:BB:7C:DF:20
a=sendrecv

2

Have you read the sticky thread on the correct forum? Asterisk General is for general discussion.