WebRTC on standalone asterisk - no audio

Question

After struggling with Asterisk for WebRTC for a few weeks now, I decided to put my problem on this forum. My Problem is as follows:

Im not getting audio from WebRTC to WebRTC clients. I work in a LAN environment.

Situation - Call from JSSIP to JSSIP (same client) with a standalone Asterisk server.

Problem

There is no audio at all when doing a call from 6001(JSSIP) to 6002(JSSIP). After a while some RTP packets are getting send, but not received. About 40 seconds after the RTP packets are showing up in the log files, the call is being terminated, without any human action.

Both clients are on the same computer, both on google chrome, were one is being runned in incognito mode. We also tested with 2 different laptops, but this gives the same results.

Does someone know what causes this issue? I also followed this thread: http://forums.digium.com/viewtopic.php?f=1&t=90167 , but this didn't worked for me. You can find the log files and further information below.

What I tried to solve this problem, due multiple posts about the same problem.

• Run Asterisk behind a Proxy (oversip)
• Used webrtc2sip, with media coding on and off
• Used SipML5 with RTCwebbreaker, disabled video
• Set asterisk public (using the WAN address and portforwarding to the server
• Run Asterisk on CentOS
• Used FireFox

Asterisk server:

• OS: Ubuntu 14.04
• Version: Asterisk 13.2 and FreePBX 12
• IP address: 192.168.178.203
• Websocket port: 8088
• SIP port: 5060

Client:

• OS: Windows 8.1
• Browser: Chrome Version 40.0.2214.115 m
• IP address: 192.168.178.18

Configurations:

rtp.conf

[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=74.125.132.127:19302

http.conf

[general]
enabled=yes
enablestatic=no
bindaddr=0.0.0.0
bindport=8088

sip.conf

[general]
port=5060
bindaddr=192.168.178.203
transport=ws,wss,udp

[6001]
host=dynamic
secret=6001
context=internal
type=friend
transport=ws,wss,udp
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=outgoing
disallow=all
allow=opus,ulaw,vp8
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/default.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=no

[6002]
host=dynamic
secret=6002
context=internal
type=friend
transport=ws,wss,udp
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=outgoing
disallow=all
allow=opus,ulaw,vp8
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/default.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=no

extensions.conf

[internal]
exten => 6001,1,Dial(SIP/6001)
exten => 6002,1,Dial(SIP/6002)

Logging

Full server log here: http://pastebin.com/qHwzXbbd

A part of the logging of the server with sip and rtp debugging on.

<------------>
Scheduling destruction of SIP dialog '0vreoff3cjblcgubthf1df' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:192.168.178.18:55387 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/WS 1t4mt0723j5t.invalid;branch=z9hG4bK7239492
Max-Forwards: 69
To: <sip:[email protected]:5060>
From: "6001" <sip:[email protected]:5060>;tag=lq9ascrlba
Call-ID: sf6m6uln2amn2lhuuvia
CSeq: 7253 INVITE
X-Can-Renegotiate: undefined
Contact: <sip:[email protected];transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.18
Content-Length: 4710

v=0
o=- 5530171873894353207 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS bEWKeIMVkeE6PTP0lgDPhtJoGfmy7MeXYx0v
m=audio 62227 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.12.1
a=rtcp:62227 IN IP4 192.168.12.1
a=candidate:3708951358 1 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3708951358 2 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3242410012 1 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:3242410012 2 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:1867667642 1 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1867667642 2 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1340102981 1 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:1340102981 2 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:2475812814 1 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2475812814 2 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2411773164 1 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:2411773164 2 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:567387210 1 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:567387210 2 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:23307701 1 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=candidate:23307701 2 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=ice-ufrag:KUz2P70p6fRKTNhU
a=ice-pwd:uSSZZuV537KDmpbtSzusDKLK
a=ice-options:google-ice
a=fingerprint:sha-256 E8:85:6A:27:CF:E0:50:A7:4B:E5:4D:25:70:76:D4:F1:5D:8D:79:93:40:4B:2E:6B:40:1D:F0:35:B9:38:56:E3
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:831281522 cname:+qS2XmtyXWZX3uFI
a=ssrc:831281522 msid:bEWKeIMVkeE6PTP0lgDPhtJoGfmy7MeXYx0v dc973b27-9b15-4cdd-bbf8-80807c4d3b1d
a=ssrc:831281522 mslabel:bEWKeIMVkeE6PTP0lgDPhtJoGfmy7MeXYx0v
a=ssrc:831281522 label:dc973b27-9b15-4cdd-bbf8-80807c4d3b1d
m=video 62227 RTP/SAVPF 100 116 117 96
c=IN IP4 192.168.12.1
a=rtcp:62227 IN IP4 192.168.12.1
a=candidate:3708951358 1 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3708951358 2 udp 2122129151 192.168.12.1 62227 typ host generation 0
a=candidate:3242410012 1 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:3242410012 2 udp 2122063615 25.124.74.215 62228 typ host generation 0
a=candidate:1867667642 1 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1867667642 2 udp 2121998079 192.168.254.1 62229 typ host generation 0
a=candidate:1340102981 1 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:1340102981 2 udp 2121932543 192.168.178.18 62230 typ host generation 0
a=candidate:2475812814 1 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2475812814 2 tcp 1518149375 192.168.12.1 0 typ host tcptype active generation 0
a=candidate:2411773164 1 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:2411773164 2 tcp 1518083839 25.124.74.215 0 typ host tcptype active generation 0
a=candidate:567387210 1 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:567387210 2 tcp 1518018303 192.168.254.1 0 typ host tcptype active generation 0
a=candidate:23307701 1 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=candidate:23307701 2 tcp 1517952767 192.168.178.18 0 typ host tcptype active generation 0
a=ice-ufrag:KUz2P70p6fRKTNhU
a=ice-pwd:uSSZZuV537KDmpbtSzusDKLK
a=ice-options:google-ice
a=fingerprint:sha-256 E8:85:6A:27:CF:E0:50:A7:4B:E5:4D:25:70:76:D4:F1:5D:8D:79:93:40:4B:2E:6B:40:1D:F0:35:B9:38:56:E3
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
<------------->
--- (15 headers 89 lines) ---
Using INVITE request as basis request - sf6m6uln2amn2lhuuvia
Found peer '6001' for '6001' from 192.168.178.18:55387

<--- Reliably Transmitting (no NAT) to 192.168.178.18:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 1t4mt0723j5t.invalid;branch=z9hG4bK7239492;received=192.168.178.18
From: "6001" <sip:[email protected]:5060>;tag=lq9ascrlba
To: <sip:[email protected]:5060>;tag=as6ed0dbe4
Call-ID: sf6m6uln2amn2lhuuvia
CSeq: 7253 INVITE
Server: FPBX-12.0.37(13.431072M)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="263201da"
Content-Length: 0
<------------>
10:57:50.828 utsx0x7f967800  ...STUN client transaction created
10:57:50.828 utsx0x7f967800  ...STUN sending message (transmit count=1)
10:57:50.828 icess0x7f970c0  ..Check 7: [2] 192.168.178.203:18141-->192.168.178.18:62230: state changed from Frozen to In Progress
10:57:50.839 utsx0x7f967800  STUN sending message (transmit count=2)
10:57:50.839 utsx0x7f967800  STUN sending message (transmit count=2)
10:57:50.850 icess0x7f970c0  Starting checklist periodic check
10:57:50.851 icess0x7f970c0  Starting checklist periodic check
10:57:50.861 utsx0x7f967800  STUN sending message (transmit count=2)
10:57:50.862 utsx0x7f967800  STUN sending message (transmit count=2)

10:57:53.770 stuse0x7f96fc0  ..TX 80 bytes STUN message to 192.168.178.18:62261:
--- begin STUN message ---
STUN Binding success response
Hdr: length=60, magic=2112a442, tsx_id=6d4d564652372f537652314a
Attributes:
  XOR-MAPPED-ADDRESS: length=8, IPv4 addr=192.168.178.18:62261
  SOFTWARE: length=12, value="pjnath-2.3.0"
  MESSAGE-INTEGRITY: length=20, data=0204b9b98f375d27e045e50bd034f1ce99288e06
  FINGERPRINT: length=4, value=1908015627 (0x71ba020b)
--- end of STUN message ---

10:57:55.586 utsx0x7f96f800  .....STUN client transaction created
10:57:55.586 utsx0x7f96f800  .....STUN sending message (transmit count=1)
10:57:55.586 icess0x7f96fc0  ....Check 7: [2] 192.168.178.203:13265-->192.168.178.18:62253: state changed from Frozen to In Progress
10:57:55.586 icess0x7f96fc0  .Performing delayed triggerred check for component 1
10:57:55.586 icess0x7f96fc0  ..Triggered check for check 6 not performed because it's in progress. Retransmitting
10:57:55.586 utsx0x7f96f801  ...STUN sending message (transmit count=1)
10:57:55.586 icess0x7f96fc0  .Performing delayed triggerred check for component 2
10:57:55.586 icess0x7f96fc0  ..Triggered check for check 7 not performed because it's in progress. Retransmitting

10:58:41.301 icess0x7f96fc0  .Triggered check for check 7 not performed because it's completed
10:58:41.301 icess0x7f96fc0  ..Check 7 is successful  and nominated
Sent RTP packet to      192.168.178.18:62249 (via ICE) (type 00, seq 020194, ts 004320, len 000160)
Sent RTP packet to      192.168.178.18:62249 (via ICE) (type 00, seq 020195, ts 004480, len 000160)
10:58:41.509 stuse0x7f970c0 !Response cache deleted
10:58:41.509 stun_session.c  tdata 0x7f96fc02cac8 destroy request, force=0, tsx=(nil)
Sent RTP packet to      192.168.178.18:62249 (via ICE) (type 00, seq 020204, ts 005920, len 000160)
10:58:41.519 stuse0x7f970c0  Response cache deleted

I hope someone can point me in some direction, or that someone has encountered the same problem and is willing to share their knowledge.

For more information, feel free to ask me.

Thank you in advance.

Solution

The configuration and the call setup looks OK. The 200 OK response would be very important to see. Also you should enable webrtc logs in the browser and check that also (or submit here) so we can see where it is actually sending the (DTLS encoded) SRTP.

Here is a detailed description about WebRTC setup in Asterisk 13.

Also it might worth to try to run asterisk on a public address (or double check all it's private ip/public ip/NAT configuration), because by default it will try to detect and use your public IP in the SIP signaling.