Asterisk 11,SIPML5,WebRTC not working in Firefox Nightly 26

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Arun Unnikrishnan

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Sep 5, 2013, 9:31:10 AM9/5/13
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I am trying to make an audio call from Firefox  to an SIP Client . Its working fine in Chrome but not in FF.


SIPML5 API version = 1.2.185
User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64; rv:26.0) Gecko/20100101 Firefox/26.0
WebSocket supported = yes
Navigator friendly name = firefox
OS friendly name = windows
Have WebRTC = yes
Have GUM = yes

I am getting the following error in Firefox :

recv=SIP/2.0 488 Not Acceptable

Any help is appreciated

 

Mamadou

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Sep 5, 2013, 9:43:05 AM9/5/13
to doub...@googlegroups.com, Arun Unnikrishnan
Firefox uses DTLS-SRTP and you'll need a server supporting such feature or webrtc2sip.org.
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Arun Unnikrishnan

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Sep 5, 2013, 9:58:40 AM9/5/13
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Sorry for not providing enough details. Currently I am trying to call from:

Browser <==>  WebRTC2SIP  <==>  Asterisk

Chrome works fine. However when I use Firefox after an INVITE is sent, I receive back from the server: SIP/2.0 488 Not Acceptable 

I have added the following in config.xml within webrtc2sip:
<enable-rtp-symetric>yes</enable-rtp-symetric>
  <enable-100rel>no</enable-100rel>
  <enable-media-coder>yes</enable-media-coder>
  <enable-videojb>no</enable-videojb>
  <video-size-pref>vga</video-size-pref>
  <rtp-buffsize>65535</rtp-buffsize>
  <avpf-tail-length>100;400</avpf-tail-length>
  <srtp-mode>optional</srtp-mode>
  <srtp-type>sdes;dtls</srtp-type>
  <dtmf-type>rfc2833</dtmf-type>
  <codecs>pcmu</codecs>
  <codec-opus-maxrates>48000;48000</codec-opus-maxrates>

I also see srtp=yes in the logs when webrtc2sip starts. What could be the reason I am getting the following back from webrtc2sip:

recv=SIP/2.0 488 Not Acceptable
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;branch=z9hG4bKKXTVpEISqhyU7BnOMwF7yt9mbr7eqR9E
From: "5004"<sip:5004@myserverip:5161>;tag=Vn6Jd3A4cgSikIN2FC6R
To: <sip:12345@myserverip>;tag=780577624
Call-ID: 8fcd898f-34ca-c1b1-8b4d-8239e3f20882
CSeq: 32875 INVITE
Content-Length: 0
Reason: text="Bad content";cause=488;text="Bad content"

Thanks is advance.

Mamadou

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Sep 5, 2013, 10:03:50 AM9/5/13
to doub...@googlegroups.com, Arun Unnikrishnan
Why do you cut the logs again? webrtc2sip logs starts with version and feature information and *we must* see the INVITE causing the 488 to understand what's the issue.
https://code.google.com/p/webrtc2sip/wiki/FAQ#How_to_report_issue?
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Arun Unnikrishnan

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Sep 5, 2013, 10:35:19 AM9/5/13
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Sorry.Here is the full log.

SIPML5 API version = 1.2.185

User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64; rv:26.0) Gecko/20100101 Firefox/26.0

WebSocket supported = yes

Navigator friendly name = firefox

OS friendly name = windows

Have WebRTC = yes

Have GUM = yes

Engine initialized

s_websocket_server_url=ws://91.xxx.xxx.xxx:10060

s_sip_outboundproxy_url=udp://91.xxx.xxx.xxx:5121

b_rtcweb_breaker_enabled=yes

b_click2call_enabled=no

SIP stack start: proxy='ns313841.ovh.net:12062', realm='<sip:91.xxx.xxx.xxx>', impi='5004', impu='"5004"<sip:50...@91.xxx.xxx.xxx:5121>'

Connecting to 'ws://91.xxx.xxx.xxx:10060'

==stack event = starting

__tsip_transport_ws_onopen

==stack event = started

State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister


SEND: REGISTER sip:91.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKZvaKbt51GTDLQiSQSWh9aueWbTAF5dpN;rport
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>
Contact: "5004"<sip:50...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1047 REGISTER
Content-Length: 0
Route: <sip:91.xxx.xxx.xxx:5121;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
Supported: path


==session event = connecting

==session event = sent_request

__tsip_transport_ws_onmessage


recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 91.xxx.xxx.xxx:10060;rport=10060;received=91.xxx.xxx.xxx;branch=z9hG4bKZvaKbt51GTDLQiSQSWh9aueWbTAF5dpN
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=as417e5e39
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1047 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP 117.241.58.80:7242;rport;branch=z9hG4bKZvaKbt51GTDLQiSQSWh9aueWbTAF5dpN;ws-hacked=WS
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="91.xxx.xxx.xxx",nonce="178c6422",stale=FALSE,algorithm=MD5


State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494


SEND: REGISTER sip:91.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKHK8F1NMQTyJHdsS0TeC6Cp7k5nE28pvX;rport
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>
Contact: "5004"<sip:50...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1048 REGISTER
Content-Length: 0
Route: <sip:91.xxx.xxx.xxx:5121;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="5004",realm="91.xxx.xxx.xxx",nonce="178c6422",uri="sip:91.xxx.xxx.xxx",response="de2a72a67772b171b8029ab3d96025ae",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom
Supported: path


==session event = sent_request

__tsip_transport_ws_onmessage


recv=OPTIONS sip:50...@91.xxx.xxx.xxx:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=117.241.58.80;ws-src-port=7242;ws-src-proto=ws SIP/2.0
Via: SIP/2.0/UDP 91.xxx.xxx.xxx:5121;rport=5161;received=91.xxx.xxx.xxx;branch=z9hG4bK07d0842a
From: "asterisk"<sip:aste...@91.xxx.xxx.xxx:5121>;tag=as796d3a07
To: <sip:50...@91.xxx.xxx.xxx:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=117.241.58.80;ws-src-port=7242;ws-src-proto=ws>
Contact: <sip:aste...@91.xxx.xxx.xxx:10060;transport=ws>
Call-ID: 4add44da49b548e5...@91.xxx.xxx.xxx:5121
CSeq: 102 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.4.0
Date: 5 Sep 2013 10:22:39 GMT;5
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer


Not implemented

SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 91.xxx.xxx.xxx:5121;rport=5161;received=91.xxx.xxx.xxx;branch=z9hG4bK07d0842a
From: "asterisk"<sip:aste...@91.xxx.xxx.xxx:5121>;tag=as796d3a07
To: <sip:50...@91.xxx.xxx.xxx:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=117.241.58.80;ws-src-port=7242;ws-src-proto=ws>
Call-ID: 4add44da49b548e5...@91.xxx.xxx.xxx:5121
CSeq: 102 OPTIONS
Content-Length: 0


__tsip_transport_ws_onmessage


recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.xxx.xxx.xxx:10060;rport=10060;received=91.xxx.xxx.xxx;branch=z9hG4bKHK8F1NMQTyJHdsS0TeC6Cp7k5nE28pvX
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=as417e5e39
Contact: <sip:50...@91.xxx.xxx.xxx:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=117.241.58.80;ws-src-port=7242;ws-src-proto=ws>;expires=200
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1048 REGISTER
Expires: 200
Content-Length: 0
Via: SIP/2.0/TCP 117.241.58.80:7242;rport;branch=z9hG4bKHK8F1NMQTyJHdsS0TeC6Cp7k5nE28pvX;ws-hacked=WS
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 5 Sep 2013 10:22:39 GMT;5


State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx

==session event = connected

State machine: c0000_Started_2_Outgoing_X_oINVITE


PeerConnectionClass = function mozRTCPeerConnection() {
    [native code]
} SessionDescriptionClass = function mozRTCSessionDescription() {
    [native code]
} IceCandidateClass = function mozRTCIceCandidate() {
    [native code]
}


ICE servers:[{"url":"stun:23.21.150.121:3478"},{"url":"stun:216.93.246.18:3478"},{"url":"stun:66.228.45.110:3478"},{"url":"stun:173.194.78.127:19302"}]

==stack event = m_permission_requested

==session event = connecting

onGetUserMediaSuccess

createOffer

==stack event = m_permission_accepted

==session event = m_stream_audio_local_added

onCreateSdpSuccess

onSetLocalDescriptionSuccess

onIceGatheringCompleted


SEND: INVITE sip:12...@91.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKUOaEWmaTqRTgzCC9W8TxnWDhrR62rZFm;rport
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=6FwXq80FZnPlD8gvdVBH
To: <sip:12...@91.xxx.xxx.xxx>
Contact: "5004"<sip:50...@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=5004;ha1=48cb589865a946cf0e7bf5455e2c6d08;+g.oma.sip-im;

+sip.ice;language="en,fr"
Call-ID: 9436a9b3-f83c-cde0-bab5-afde1d86fc2b
CSeq: 4378 INVITE
Content-Type: application/sdp
Content-Length: 1337
Route: <sip:91.xxx.xxx.xxx:5121;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom

v=0
o=Mozilla-SIPUA-26.0a1 2966 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:feb20971
a=ice-pwd:e2db996b527306b7566f9db2fb96c29a
a=fingerprint:sha-256 69:76:A9:97:80:80:F2:7C:35:C6:12:1F:21:C2:DA:AA:10:86:77:21:BE:77:3A:D9:3A:30:67:01:F8:4F:31:A7
m=audio 54472 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 117.241.58.80
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:0 1 UDP 2128609535 192.168.2.7 54472 typ host
a=candidate:1 1 UDP 1692467199 117.241.58.80 54472 typ srflx raddr 192.168.2.7 rport 54472
a=candidate:2 1 UDP 1692467199 117.241.58.80 1041 typ srflx raddr 192.168.2.7 rport 54472
a=candidate:3 1 UDP 1692467199 117.241.58.80 1110 typ srflx raddr 192.168.2.7 rport 54472
a=candidate:4 1 UDP 1692467199 117.241.58.80 1119 typ srflx raddr 192.168.2.7 rport 54472
a=candidate:0 2 UDP 2128609534 192.168.2.7 54473 typ host
a=candidate:1 2 UDP 1692467198 117.241.58.80 54473 typ srflx raddr 192.168.2.7 rport 54473
a=candidate:2 2 UDP 1692467198 117.241.58.80 1120 typ srflx raddr 192.168.2.7 rport 54473
a=candidate:3 2 UDP 1692467198 117.241.58.80 1121 typ srflx raddr 192.168.2.7 rport 54473
a=candidate:4 2 UDP 1692467198 117.241.58.80 1122 typ srflx raddr 192.168.2.7 rport 54473


onIceCandidate = undefined

ICE GATHERING COMPLETED!

onIceGatheringCompleted

onIceGatheringCompleted but no local sdp request is pending

__tsip_transport_ws_onmessage


recv=SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;branch=z9hG4bKUOaEWmaTqRTgzCC9W8TxnWDhrR62rZFm
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=6FwXq80FZnPlD8gvdVBH
To: <sip:12...@91.xxx.xxx.xxx>
Call-ID: 9436a9b3-f83c-cde0-bab5-afde1d86fc2b
CSeq: 4378 INVITE
Content-Length: 0


State machine: x0000_Any_2_Any_X_i1xx

==session event = i_ao_request

__tsip_transport_ws_onmessage


recv=SIP/2.0 488 Not Acceptable
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;branch=z9hG4bKUOaEWmaTqRTgzCC9W8TxnWDhrR62rZFm
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=6FwXq80FZnPlD8gvdVBH
To: <sip:12...@91.xxx.xxx.xxx>;tag=839627435
Call-ID: 9436a9b3-f83c-cde0-bab5-afde1d86fc2b
CSeq: 4378 INVITE
Content-Length: 0
Reason: text="Bad content";cause=488;text="Bad content"



SEND: ACK sip:12...@91.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKUOaEWmaTqRTgzCC9W8TxnWDhrR62rZFm;rport
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=6FwXq80FZnPlD8gvdVBH
To: <sip:12...@91.xxx.xxx.xxx>;tag=839627435
Call-ID: 9436a9b3-f83c-cde0-bab5-afde1d86fc2b
CSeq: 4378 ACK
Content-Length: 0
Route: <sip:91.xxx.xxx.xxx:5121;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70


State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE

=== INVITE Dialog terminated ===

PeerConnection::stop()

==session event = i_ao_request

==session event = terminated

The FSM is in the final state

State machine: tsip_dialog_register_Any_2_InProgress_X_shutdown


SEND: REGISTER sip:91.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKhgWoyx34X6zGnFLQi10gTbLuX4EhFk9K;rport
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>
Contact: "5004"<sip:50...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=0;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1049 REGISTER
Content-Length: 0
Route: <sip:91.xxx.xxx.xxx:5121;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="5004",realm="91.xxx.xxx.xxx",nonce="178c6422",uri="sip:91.xxx.xxx.xxx",response="de2a72a67772b171b8029ab3d96025ae",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom


==session event = terminating

==session event = sent_request

__tsip_transport_ws_onmessage


recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 91.xxx.xxx.xxx:10060;rport=10060;received=91.xxx.xxx.xxx;branch=z9hG4bKhgWoyx34X6zGnFLQi10gTbLuX4EhFk9K
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=as417e5e39
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1049 REGISTER
Content-Length: 0
Via: SIP/2.0/TCP 117.241.58.80:7242;rport;branch=z9hG4bKhgWoyx34X6zGnFLQi10gTbLuX4EhFk9K;ws-hacked=WS
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="91.xxx.xxx.xxx",nonce="1e71b126",stale=TRUE,algorithm=MD5


State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494


SEND: REGISTER sip:91.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKCTGnT45hkNnlviPfo5nc8M5dV5SalJeT;rport
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>
Contact: "5004"<sip:50...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=0;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1050 REGISTER
Content-Length: 0
Route: <sip:91.xxx.xxx.xxx:5121;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="5004",realm="91.xxx.xxx.xxx",nonce="1e71b126",uri="sip:91.xxx.xxx.xxx",response="50c79549bb7b35321bcf5f93f06662f5",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.07.17
Organization: Doubango Telecom


==session event = sent_request

__tsip_transport_ws_onmessage


recv=SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.xxx.xxx.xxx:10060;rport=10060;received=91.xxx.xxx.xxx;branch=z9hG4bKCTGnT45hkNnlviPfo5nc8M5dV5SalJeT
From: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=zBW5NEtrKISKT0BG2pEO
To: "5004"<sip:50...@91.xxx.xxx.xxx:5121>;tag=as417e5e39
Contact: <sip:50...@91.xxx.xxx.xxx:10060;rtcweb-breaker=yes;transport=udp;ws-src-ip=117.241.58.80;ws-src-port=7242;ws-src-proto=ws>;expires=200
Call-ID: 7399bd17-6e4a-e30b-cb4a-1a0d404d291b
CSeq: 1050 REGISTER
Expires: 200
Content-Length: 0
Via: SIP/2.0/TCP 117.241.58.80:7242;rport;branch=z9hG4bKCTGnT45hkNnlviPfo5nc8M5dV5SalJeT;ws-hacked=WS
Server: Asterisk PBX 11.4.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 5 Sep 2013 10:22:43 GMT;5


State machine: tsip_dialog_register_InProgress_2_Terminated_X_2xx

=== REGISTER Dialog terminated ===

==session event = terminated

==stack event = stopped

__tsip_transport_ws_onclose

==stack event = stopped


On Thursday, September 5, 2013 7:01:10 PM UTC+5:30, Arun Unnikrishnan wrote:

Mamadou

unread,
Sep 5, 2013, 10:41:21 AM9/5/13
to doub...@googlegroups.com, Arun Unnikrishnan
Chrome uses SRTP-SDES while Firefox uses SRTP-DTLS if you shared the logs we would probably see errors about DTLS not being enabled.
You have to provide SSL certificates for SRTP-DTLS to work: https://code.google.com/p/webrtc2sip/wiki/FAQ#I_see_"Remote_party_requesting_DTLS-DTLS_(UDP/TLS/RTP/SAVPF
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Arun Unnikrishnan

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Sep 6, 2013, 10:16:22 PM9/6/13
to doub...@googlegroups.com
Thanks Mamadou. You are right.SSL certificate was missing .Added  self signed ssl certificate and now its working fine in Firefox .Thanks again for your support and help. 


On Thursday, September 5, 2013 7:01:10 PM UTC+5:30, Arun Unnikrishnan wrote:
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