[Spce-user] Web RTC error

Wed Jul 8 15:19:58 EDT 2015

Trying to get things working and am stumbling.  Maybe someone can help me a
bit?

Right now, I just want to get things working, ie do a simple test using
jssip.net, based on the AWS AMI image built by sipwise, ie sip:provider CE
AMI mr3.8.2, image id:  ami-17142e27 (us west 2)

Here's my steps so far:

1) got the ec2 instance running
2) configured the ec2 security group/ports as below:

HTTP
  TCP
  80
  0.0.0.0/0
  HTTPS
  TCP
  443
  0.0.0.0/0
  Custom TCP Rule
  TCP
  1080
  0.0.0.0/0
  Custom TCP Rule
  TCP
  1443
  0.0.0.0/0
  Custom TCP Rule
  TCP
  2443
  0.0.0.0/0
  Custom TCP Rule
  TCP
  5060
  0.0.0.0/0
  Custom TCP Rule
  TCP
  5061
  0.0.0.0/0
  Custom UDP Rule
  UDP
  5060
  0.0.0.0/0
  Custom UDP Rule
  UDP
  5061
  0.0.0.0/0
ssh is configured for my machine only (obviously)

3) got a proper ssl cert from godaddy, change all my sslcerfile and sslkey
files in config.yml appropriately, and made sure kamailio tls is enabled
(which it is by default in the ami) ran ngcpcfg apply  (everything was
happy so far).

4) launched firefox under linux going to tryit.jssip.net, with folowing
details:
name:  quandt
sip uri:  sip:quandt at sip.autodcp.com
password:  ******
ws uri:  wss://sip.autodcp.com:1443/wss/sip/

Which got me to the jssip demo page both connected and registered just fine.

5) on a mac launched zoiper and logged into another account on my sip server

6) tried to call from one to the other.  Got a ring from one ot the other
to work, on the jssip demo page, when I ansewred, I get promoted to share
my microphone, which I acknowlege, and them get a WebRTC error right
away.   Below is part of the console messages.

Any thoughts?

Yours truly,
Brian

" " +2s" jssip.js:21621
"JsSIP:Transport " "sending WebSocket message:

SIP/2.0 200 OK

Via: SIP/2.0/WSS 54.189.6.185:5061
;branch=z9hG4bKdf8e.4e2788dc9d9e3774bc01623f84011d82.0

Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0

To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws;tag=je09o4s8o3

From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-8205b6a6

Call-ID: 8c334a51-c57f0854-1af23e2 at 127.0.0.1

CSeq: 1 OPTIONS

Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS

Accept: application/sdp, application/dtmf-relay

Supported: outbound

Content-Length: 0

" " +16ms" jssip.js:21621
"JsSIP:NonInviteServerTransaction " "Timer J expired for transaction
z9hG4bKdf8e.4e2788dc9d9e3774bc01623f84011d82.0" " +3ms" jssip.js:21621
"JsSIP:RTCSession " "answer()" " +1s" jssip.js:21621
"JsSIP:Dialog " "dialog
MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700
changed to CONFIRMED state" " +0ms" jssip.js:21621
"rtcninja:RTCPeerConnection " "new | pcConfig: " Object { iceServers:
Array[1], gatheringTimeout: 2000 } " +3ms" jssip.js:21621
"rtcninja:RTCPeerConnection " "setConfigurationAndOptions | processed
pcConfig: " Object { iceServers: Array[1] } " +1ms" jssip.js:21621
"rtcninja:Adapter " "getUserMedia() | constraints: " Object { audio: true,
video: false } " +93ms" jssip.js:21621
Invalid URI. Load of media resource  failed. tryit.jssip.net
"rtcninja:Adapter " "getUserMedia() | success" " +2s" jssip.js:21621
"rtcninja:RTCPeerConnection " "addStream() | stream: [object
LocalMediaStream]" " +0ms" jssip.js:21621
"rtcninja:RTCPeerConnection " "setRemoteDescription()" " +1ms"
jssip.js:21621
"rtcninja:ERROR:RTCPeerConnection " "setRemoteDescription() | error:" "
+1ms" Object { name: "INVALID_SESSION_DESCRIPTION", message: "Could not
negotiate media lines; cause = NO_DTLS_FINGERPRINT | SDP Parsing Error:
Warning: No network type specified in comediadir attribute.",
__exposedProps__: Object } jssip.js:21796
"JsSIP:Transport " "sending WebSocket message:

SIP/2.0 488 Not Acceptable Here

Via: SIP/2.0/WSS 54.189.6.185:5061
;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0

Via: SIP/2.0/UDP 127.0.0.1:5080
;received=127.0.0.1;branch=z9hG4bKuN~pJa9J;rport=5080

To: <sip:d62a2g56 at sip.autodcp.com>;tag=2mftbnm4nm

From: <sip:0991002 at sip.autodcp.com>;tag=5A6A45EC-559D727B0008760D-D4D52700

Call-ID: MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-1

CSeq: 10 INVITE

Supported: timer,ice,outbound

Content-Length: 0

" " +0ms" jssip.js:21621
"JsSIP:RTCSession " "session failed" " +1ms" jssip.js:21621
"JsSIP:RTCSession " "close()" " +0ms" jssip.js:21621
"rtcninja:RTCPeerConnection " "close()" " +0ms" jssip.js:21621
"rtcninja:RTCPeerConnection " "oniceconnectionstatechange() |
iceConnectionState: closed" " +0ms" jssip.js:21621
"JsSIP:RTCSession " "close() | closing local MediaStream" " +0ms"
jssip.js:21621
"rtcninja:Adapter " "closeMediaStream() | calling stop() on all the
MediaStreamTrack" " +1ms" jssip.js:21621
"JsSIP:Dialog " "dialog
MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700
deleted" " +4ms" jssip.js:21621
"rtcninja:RTCPeerConnection " "onsignalingstatechange() | signalingState:
closed" " +5ms" jssip.js:21621
"JsSIP:Transport " "received WebSocket text message:

ACK sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws SIP/2.0

Max-Forwards: 70

Record-Route: <sip:54.189.6.185:5060
;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061
;ftag=5A6A45EC-559D727B0008760D-D4D52700;lr=on>

Record-Route: <sip:127.0.0.1:5060
;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061
;ftag=5A6A45EC-559D727B0008760D-D4D52700;lr=on>

Via: SIP/2.0/WSS 54.189.6.185:5061
;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0

Via: SIP/2.0/UDP 127.0.0.1:5080
;received=127.0.0.1;branch=z9hG4bKuN~pJa9J;rport=5080

From: <sip:0991002 at sip.autodcp.com>;tag=5A6A45EC-559D727B0008760D-D4D52700

To: <sip:d62a2g56 at sip.autodcp.com>;tag=2mftbnm4nm

Call-ID: MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-1

CSeq: 10 ACK

Content-Length: 0

Route: <sip:10.220.196.230:32769;transport=ws>

" " +31ms" jssip.js:21621
"JsSIP:Transport " "received WebSocket text message:

OPTIONS sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws SIP/2.0

Max-Forwards: 70

Record-Route: <sip:54.189.6.185:5060
;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061
;ftag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6;lr=on>

Record-Route: <sip:127.0.0.1:5060
;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061
;ftag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6;lr=on>

Via: SIP/2.0/WSS 54.189.6.185:5061
;branch=z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0

Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0

From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6

To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws

Call-ID: 8c334a51-067f0854-fbf23e2 at 127.0.0.1

CSeq: 1 OPTIONS

Content-Length: 0

" " +27s" jssip.js:21621
"JsSIP:Transport " "sending WebSocket message:

SIP/2.0 200 OK

Via: SIP/2.0/WSS 54.189.6.185:5061
;branch=z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0

Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0

To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws;tag=5r8pi0ggps

From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6

Call-ID: 8c334a51-067f0854-fbf23e2 at 127.0.0.1

CSeq: 1 OPTIONS

Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS

Accept: application/sdp, application/dtmf-relay

Supported: outbound

Content-Length: 0

" " +7ms" jssip.js:21621
"JsSIP:NonInviteServerTransaction " "Timer J expired for transaction
z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0" " +0ms" jssip.js:21621
"JsSIP:InviteServerTransaction " "Timer H expired for transaction
z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0" " +5s" jssip.js:21621
"JsSIP:Transport " "received WebSocket text message:

OPTIONS sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws SIP/2.0

Max-Forwards: 70

Record-Route: <sip:54.189.6.185:5060
;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061
;ftag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6;lr=on>

Record-Route: <sip:127.0.0.1:5060
;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061
;ftag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6;lr=on>

Via: SIP/2.0/WSS 54.189.6.185:5061
;branch=z9hG4bK68e6.7c6248942a593cf21da989d27e9b4cf3.0

Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0

From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6

To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws

Call-ID: 8c334a51-467f0854-ddf23e2 at 127.0.0.1

CSeq: 1 OPTIONS

Content-Length: 0

" " +25s" jssip.js:21621
"JsSIP:Transport " "sending WebSocket message:

SIP/2.0 200 OK

Via: SIP/2.0/WSS 54.189.6.185:5061
;branch=z9hG4bK68e6.7c6248942a593cf21da989d27e9b4cf3.0

Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0

To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws;tag=0vv3pidftj

From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6

Call-ID: 8c334a51-467f0854-ddf23e2 at 127.0.0.1

CSeq: 1 OPTIONS

Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS

Accept: application/sdp, application/dtmf-relay

Supported: outbound

Content-Length: 0
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