title: Asterisk gives "Strict RTP learning" message and no audio for Chrome WebRTC but works in Firefox
body: <p>I've been experimenting with WebRTC with an Asterisk server (v13.18) on the same LAN as my computer.  I configured the Asterisk extension 6003 to automatically answer and play a certain notorious sound file whenever it's dialed, then confirmed that this worked with the Ekiga softphone client.</p>
  
  <p>I was then able to get this working as well in Firefox via the following steps:</p>
  
  <ul>
  <li>Opened the online demo site <a href="https://www.doubango.org/sipml5/call.htm?svn=252" rel="noreferrer">https://www.doubango.org/sipml5/call.htm?svn=252</a></li>
  <li>Opened the Expert Mode screen</li>
  <li>Checked the "Disable Video" checkbox.</li>
  <li>Entered <code>[]</code> for the "ICE servers" field (because I'm on a local LAN with no NAT involved, I don't need STUN or TURN, though I do have ICE enabled in my Asterisk config)</li>
  <li>Entered my "Websocket Server URL" value of <code>wss://asterisk-ci.test:8089/ws</code></li>
  <li>Clicked Save and then returned to the demo page.</li>
  <li>Entered the asterisk server info on the "Registration" section of the demo page and clicked "Login", confirming that it then displays a status of "Connected".</li>
  <li>Entered the extension 6003 in the "Call Control" box and clicked "Call".</li>
  </ul>
  
  <p>In Firefox this works great - the sound file is played back to me over the call.</p>
  
  <p>In Google Chrome (latest v65) no sound actually plays but other than that everything seems like it should be working.  In particular:</p>
  
  <ul>
  <li>The sipML5 client displays "In Call" and the UI shows the call being active.</li>
  <li>No errors in the Javascript console.</li>
  <li>The SIP traffic in the Websocket Frames in the Network tab look good and seem to match what Firefox is doing.</li>
  <li>The <code>chrome://webrtc-internals</code> page indicates a lot of traffic coming in.  In particular, the graph of the data on the audio channel appears consistent with a sound file showing up here.</li>
  </ul>
  
  <p>I tried setting up an example app using <code>SIP.js</code> and got the exact same results, confirming that this isn't an issue with <code>sipML5</code> but is rather something about my Asterisk config and how it interacts with Google Chrome.</p>
  
  <p>I connected to Asterisk via <code>asterisk -vvvvvr</code> to see what debug messages might show me, and there do appear to be some significant differences between the working Firefox and the nonworking Chrome.  Here's what I see in Firefox when connecting and then making the call:</p>
  
  <pre><code>== WebSocket connection from '192.168.99.123:40190' for protocol 'sip' accepted using version '13'
    -- Registered SIP '1061' at 192.168.99.123:40190
  == DTLS ECDH initialized (automatic), faster PFS enabled
  == Using SIP RTP CoS mark 5
       &gt; 0x7f79f800dba0 -- Strict RTP learning after remote address set to: 192.168.99.123:32807
    -- Executing [6003@users:1] Answer("SIP/1061-00000007", "") in new stack
       &gt; 0x7f79f800dba0 -- Strict RTP learning after ICE completion
       &gt; 0x7f79f800dba0 -- Strict RTP switching to RTP target address 192.168.99.123:32807 as source
    -- Executing [6003@users:2] Playback("SIP/1061-00000007", "auto-playback") in new stack
    -- &lt;SIP/1061-00000007&gt; Playing 'auto-playback.slin' (language 'en')
       &gt; 0x7f79f800dba0 -- Strict RTP learning complete - Locking on source address 192.168.99.123:32807
    -- Executing [6003@users:3] Hangup("SIP/1061-00000006", "") in new stack
  </code></pre>
  
  <p>But I get a very different result when connecting on Google Chrome:</p>
  
  <pre><code>== WebSocket connection from '192.168.99.123:52868' for protocol 'sip' accepted using version '13'
    -- Registered SIP '1061' at 192.168.99.123:52868
  == DTLS ECDH initialized (automatic), faster PFS enabled
  == Using SIP RTP CoS mark 5
       &gt; 0x7f79fc00c710 -- Strict RTP learning after remote address set to: 127.0.0.1:9
    -- Executing [6003@users:1] Answer("SIP/1061-00000008", "") in new stack
       &gt; 0x7f79fc00c710 -- Strict RTP learning after remote address set to: 192.168.99.123:39303
       &gt; 0x7f79fc00c710 -- Strict RTP learning after remote address set to: 192.168.99.123:39303
       &gt; 0x7f79fc00c710 -- Strict RTP learning after remote address set to: 192.168.99.123:39303
    -- Executing [6003@users:2] Playback("SIP/1061-00000008", "auto-playback") in new stack
    -- &lt;SIP/1061-00000008&gt; Playing 'auto-playback.slin' (language 'en')
       &gt; 0x7f79fc00c710 -- Strict RTP learning after remote address set to: 192.168.99.123:39303
       &gt; 0x7f79fc00c710 -- Strict RTP learning after remote address set to: 192.168.99.123:39303
  </code></pre>
  
  <p>The message <code>0x7f79fc00c710 -- Strict RTP learning after remote address set to: 192.168.99.123:39303</code> then repeats ad infinitum for the duration of the call.</p>
  
  <p>In addition to that message repeating, I notice that on Firefox, the original "Strict RTP learning" message has the correct address, whereas on Google Chrome is has <code>127.0.0.1:9</code>.  Both the <code>127.0.0.1</code> and the use of port <code>9</code> is interesting, though I'm not sure what to make of either.  Does Google Chrome hide your IP address in a way which is messing with Asterisk?</p>
  
  <p>Interestingly, when I try the same thing using a SIP.js example, I get exactly the same result (works on Firefox, connects but has no sound on Chrome) with the same debug output in Asterisk except that the initial address is <code>0.0.0.0:9</code> instead of <code>127.0.0.1:9</code>.</p>
  
  <p>Regardless I'm not sure what next steps to even try, so any help would be appreciated.</p>
  
  <p><strong>EDIT:</strong> As suggested, I'll post the SDP logs.  Here's what I get for the working Firefox:</p>
  
  <pre><code>Local SDP (Offer)
  v=0
  o=mozilla...THIS_IS_SDPARTA-59.0.2 7697709853700369104 0 IN IP4 0.0.0.0
  s=-
  t=0 0
  a=sendrecv
  a=fingerprint:sha-256 BD:03:D7:1A:FB:F7:A3:BE:D0:F9:22:65:80:7B:FE:78:1C:17:01:17:99:57:A4:40:49:0D:EF:AA:AA:91:63:2C
  a=group:BUNDLE sdparta_0
  a=ice-options:trickle
  a=msid-semantic:WMS *
  m=audio 52547 UDP/TLS/RTP/SAVPF 109 9 0 8 101
  c=IN IP4 192.168.99.123
  a=candidate:0 1 UDP 2122252543 192.168.99.123 52547 typ host
  a=candidate:1 1 TCP 2105524479 192.168.99.123 9 typ host tcptype active
  a=candidate:0 2 UDP 2122252542 192.168.99.123 33797 typ host
  a=candidate:1 2 TCP 2105524478 192.168.99.123 9 typ host tcptype active
  a=sendrecv
  a=end-of-candidates
  a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:mid
  a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
  a=fmtp:101 0-15
  a=ice-pwd:63350c71006d1daf78366efc8d05347f
  a=ice-ufrag:e92ccf7b
  a=mid:sdparta_0
  a=msid:{8a0a921d-b591-41b5-94e7-647b9b40cd06} {78e4a3a8-628f-4e09-a05a-fa6edb3022be}
  a=rtcp:33797 IN IP4 192.168.99.123
  a=rtcp-mux
  a=rtpmap:109 opus/48000/2
  a=rtpmap:9 G722/8000/1
  a=rtpmap:0 PCMU/8000
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=setup:actpass
  a=ssrc:1153204890 cname:{9de8930f-bf76-48e0-a9c9-4c15f6914409}
  Remote SDP (Answer)
  v=0
  o=root 477460967 477460967 IN IP4 172.30.8.8
  s=-
  t=0 0
  a=sendrecv
  m=audio 18666 RTP/SAVPF 0 8 101
  c=IN IP4 172.30.8.8
  a=candidate:Hac1e0808 1 UDP 2130706431 172.30.8.8 18666 typ host
  a=candidate:Hac1e0808 2 UDP 2130706430 172.30.8.8 18667 typ host
  a=sendrecv
  a=fingerprint:sha-256 75:D2:BE:77:B6:8E:1B:4E:F9:BF:FB:34:54:2D:05:31:F6:97:C5:34:F3:D9:65:BE:FC:C6:E4:5C:1A:5E:11:E7
  a=fmtp:101 0-16
  a=ice-pwd:1e0f3ac370cce57d7c978ecb57ae23d9
  a=ice-ufrag:7189ea580175062f339a9fe84ed6ecae
  a=maxptime:150
  a=rtcp-mux
  a=rtpmap:0 PCMU/8000
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=setup:active
  </code></pre>
  
  <p>And here's what I see from the non-working Chrome, which also looks like SDP but looks so different than I e.g. don't even see my IP address in any of the output:</p>
  
  <pre><code>&gt; createOfferOnSuccess
  type: offer, sdp: v=0
  o=- 3202047122122577027 2 IN IP4 127.0.0.1
  s=-
  t=0 0
  a=group:BUNDLE audio
  a=msid-semantic: WMS C0FThsSoaGKxFOoR8Fnptw8vJWdbuN4K2DeU
  m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  c=IN IP4 0.0.0.0
  a=rtcp:9 IN IP4 0.0.0.0
  a=ice-ufrag:kQT+
  a=ice-pwd:6BgMZ48o3m7PMPFXY7AZvfdb
  a=ice-options:trickle
  a=fingerprint:sha-256 59:9F:B3:53:89:64:3A:3F:03:1B:32:8F:97:9B:6E:A1:33:B8:05:DD:92:87:3C:1C:CA:A3:83:28:8D:2C:98:FE
  a=setup:actpass
  a=mid:audio
  a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  a=sendrecv
  a=rtcp-mux
  a=rtpmap:111 opus/48000/2
  a=rtcp-fb:111 transport-cc
  a=fmtp:111 minptime=10;useinbandfec=1
  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:110 telephone-event/48000
  a=rtpmap:112 telephone-event/32000
  a=rtpmap:113 telephone-event/16000
  a=rtpmap:126 telephone-event/8000
  a=ssrc:3990625320 cname:R43Nh5Jptx9sDbOE
  a=ssrc:3990625320 msid:C0FThsSoaGKxFOoR8Fnptw8vJWdbuN4K2DeU c42217e8-19c2-4d94-a392-d4166d00eb22
  a=ssrc:3990625320 mslabel:C0FThsSoaGKxFOoR8Fnptw8vJWdbuN4K2DeU
  a=ssrc:3990625320 label:c42217e8-19c2-4d94-a392-d4166d00eb22
  
  &gt; setLocalDescription
  type: offer, sdp: v=0
  o=- 3202047122122577027 2 IN IP4 127.0.0.1
  s=-
  t=0 0
  a=group:BUNDLE audio
  a=msid-semantic: WMS C0FThsSoaGKxFOoR8Fnptw8vJWdbuN4K2DeU
  m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  c=IN IP4 0.0.0.0
  a=rtcp:9 IN IP4 0.0.0.0
  a=ice-ufrag:kQT+
  a=ice-pwd:6BgMZ48o3m7PMPFXY7AZvfdb
  a=ice-options:trickle
  a=fingerprint:sha-256 59:9F:B3:53:89:64:3A:3F:03:1B:32:8F:97:9B:6E:A1:33:B8:05:DD:92:87:3C:1C:CA:A3:83:28:8D:2C:98:FE
  a=setup:actpass
  a=mid:audio
  a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  a=sendrecv
  a=rtcp-mux
  a=rtpmap:111 opus/48000/2
  a=rtcp-fb:111 transport-cc
  a=fmtp:111 minptime=10;useinbandfec=1
  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:110 telephone-event/48000
  a=rtpmap:112 telephone-event/32000
  a=rtpmap:113 telephone-event/16000
  a=rtpmap:126 telephone-event/8000
  a=ssrc:3990625320 cname:R43Nh5Jptx9sDbOE
  a=ssrc:3990625320 msid:C0FThsSoaGKxFOoR8Fnptw8vJWdbuN4K2DeU c42217e8-19c2-4d94-a392-d4166d00eb22
  a=ssrc:3990625320 mslabel:C0FThsSoaGKxFOoR8Fnptw8vJWdbuN4K2DeU
  a=ssrc:3990625320 label:c42217e8-19c2-4d94-a392-d4166d00eb22
  
  &gt; setRemoteDescription
  type: answer, sdp: v=0
  o=root 2070370846 2070370846 IN IP4 172.30.8.8
  s=Asterisk PBX certified/13.18-cert2
  c=IN IP4 172.30.8.8
  t=0 0
  m=audio 13528 RTP/SAVPF 0 8 126
  a=rtpmap:0 PCMU/8000
  a=rtpmap:8 PCMA/8000
  a=rtpmap:126 telephone-event/8000
  a=fmtp:126 0-16
  a=maxptime:150
  a=ice-ufrag:4c551f814a951c5e6f74e5c225c5e160
  a=ice-pwd:7877be1235781d443361467a70b33c12
  a=candidate:Hac1e0808 1 UDP 2130706431 172.30.8.8 13528 typ host
  a=candidate:Hac1e0808 2 UDP 2130706430 172.30.8.8 13529 typ host
  a=connection:new
  a=setup:active
  a=fingerprint:SHA-256 75:D2:BE:77:B6:8E:1B:4E:F9:BF:FB:34:54:2D:05:31:F6:97:C5:34:F3:D9:65:BE:FC:C6:E4:5C:1A:5E:11:E7
  a=rtcp-mux
  a=sendrecv
  </code></pre>
  
  <p>And in case it helps, here's the complete set of events logged when setting up the call and playing it:</p>
  
  <pre><code>addStream
  createOffer
  negotiationneeded
  createOfferOnSuccess
  setLocalDescription
  signalingstatechange
  setLocalDescriptionOnSuccess
  icegatheringstatechange
  icegatheringstatechange
  setRemoteDescription
  signalingstatechange
  iceconnectionstatechange
  onAddStream
  setRemoteDescriptionOnSuccess
  </code></pre>
  
  <p><strong>FURTHER EDIT:</strong> After reviewing some <a href="https://webrtchacks.com/sdp-anatomy/" rel="noreferrer">SDP docs</a> and looking through my own SDP logs above, the main thing I see that differs and probably accounts for Firefox working and Chrome not is that Firefox has the line</p>
  
  <pre><code>c=IN IP4 192.168.99.123
  </code></pre>
  
  <p>which is indeed my IP address, whereas Chrome has the line</p>
  
  <pre><code>c=IN IP4 0.0.0.0
  </code></pre>
  
  <p>I tried running Chrome from a terminal to capture any debug output that gets printed to the screen apart from what I see in chrome://webrtc-internals and I found that this message is displayed many times per second:</p>
  
  <pre><code>ERROR:dtlstransport.cc(557)] Jingle:DtlsTransport[audio|1|__]: Received non-DTLS packet before DTLS complete.
  </code></pre>
  
  <p>I've read through a number of Google search results for that error but haven't been able to come up with anything to try to fix it.  However, it seems possibly related; if one or more UDP packets went to the wrong place, then even if most of the audio packets were properly sent, they'd never get decoded and we'd see a lot of data coming in but no audio actually being played.  Which is indeed what I'm seeing.</p>
  
  <p>I'll do some more digging to see what settings I can tweak to make Chrome send the same sort of information that Firefox is sending, or to have Asterisk do the correct thing for both of them.  In the meantime, I'm opening a Bounty on this question, since any additional help and suggestions will be much appreciated.</p>