No audio for SIPml5 to softphone calling via asterisk

milind_r · March 20, 2014, 12:32pm

hi…when i play sound file from SIPml using the Playback() application then there is no issue of audio…but when i call from crome based SIPml5 client to zoiper sipphone then call is get connected but there is no audio i attache the sipml5 crome based console log and asterisk cli log for dubug.i install doubango framework as well as webrtc2sip on server[code]SIPML5 API version = 1.3.214 SIPml-api.js?svn=179:1
location=http://sipml5.org/call.htm?svn=203# call.htm?svn=203:147
User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/33.0.1750.154 Safari/537.36 SIPml-api.js?svn=179:1
WebSocket supported = yes SIPml-api.js?svn=179:1
Navigator friendly name = chrome SIPml-api.js?svn=179:1
OS friendly name = windows SIPml-api.js?svn=179:1
Have WebRTC = yes SIPml-api.js?svn=179:1
Have GUM = yes SIPml-api.js?svn=179:1
Engine initialized SIPml-api.js?svn=179:1
event.returnValue is deprecated. Please use the standard event.preventDefault() instead. jquery.js:2
s_websocket_server_url=ws://202.141.151.35:8088/ws SIPml-api.js?svn=179:1
s_sip_outboundproxy_url=(null) SIPml-api.js?svn=179:1
b_rtcweb_breaker_enabled=yes SIPml-api.js?svn=179:1
b_click2call_enabled=no SIPml-api.js?svn=179:1
b_early_ims=no SIPml-api.js?svn=179:1
b_enable_media_stream_cache=yes SIPml-api.js?svn=179:1
o_bandwidth={} SIPml-api.js?svn=179:1
o_video_size={} SIPml-api.js?svn=179:1
SIP stack start: proxy=‘ns313841.ovh.net:13062’, realm=‘sip:202.141.151.35’, impi=‘1060’, impu=’"milind"sip:1060@202.141.151.35’ SIPml-api.js?svn=179:1
Connecting to ‘ws://202.141.151.35:8088/ws’ SIPml-api.js?svn=179:1
==stack event = starting SIPml-api.js?svn=179:1
__tsip_transport_ws_onopen SIPml-api.js?svn=179:1
==stack event = started SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister SIPml-api.js?svn=179:1
SEND: REGISTER sip:202.141.151.35 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKCz5395ozGLWvLRXOb2l9zntym6bCzNGf;rport
From: "milind"sip:1060@202.141.151.35;tag=uAd1ZHfTlZbvQyIzT8CT
To: “milind"sip:1060@202.141.151.35
Contact: “milind"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language=“en,fr"
Call-ID: 76738299-e0fd-8ba5-df46-ef4a0cf6e0f8
CSeq: 7931 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“202.141.151.35”,nonce=””,uri=“sip:202.141.151.35”,response=”"
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.10
Organization: Doubango Telecom
Supported: path

SIPml-api.js?svn=179:1
==session event = connecting SIPml-api.js?svn=179:1
==session event = sent_request SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=64239;received=202.141.151.100;branch=z9hG4bKCz5395ozGLWvLRXOb2l9zntym6bCzNGf
From: "milind"sip:1060@202.141.151.35;tag=uAd1ZHfTlZbvQyIzT8CT
To: "milind"sip:1060@202.141.151.35;tag=as6900d7f4
Call-ID: 76738299-e0fd-8ba5-df46-ef4a0cf6e0f8
CSeq: 7931 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.7.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm=“202.141.151.35”,nonce=“5e823488”,stale=FALSE,algorithm=MD5

SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 SIPml-api.js?svn=179:1
SEND: REGISTER sip:202.141.151.35 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKqnSaIisdOMSHpOCLBpGd4koXFtz7WPB4;rport
From: "milind"sip:1060@202.141.151.35;tag=uAd1ZHfTlZbvQyIzT8CT
To: "milind"sip:1060@202.141.151.35
Contact: "milind"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 76738299-e0fd-8ba5-df46-ef4a0cf6e0f8
CSeq: 7932 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“202.141.151.35”,nonce=“5e823488”,uri=“sip:202.141.151.35”,response=“45e4e5efb08ac46137b43e7ec84e0278”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.10
Organization: Doubango Telecom
Supported: path

SIPml-api.js?svn=179:1
==session event = sent_request SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=OPTIONS sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws SIP/2.0
Via: SIP/2.0/WS 202.141.151.35:5060;rport;branch=z9hG4bK7551b3d6
From: "asterisk"sip:asterisk@202.141.151.35;tag=as382c9b91
To: sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws
Contact: sip:asterisk@202.141.151.35:5060;transport=WS
Call-ID: 68ac7e7f38cbf60f4bc424c9167d1429@202.141.151.35:5060
CSeq: 102 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX 11.7.0
Date: 20 Mar 2014 12:23:02 GMT;20
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

SIPml-api.js?svn=179:1
Not implemented SIPml-api.js?svn=179:1
SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 202.141.151.35:5060;rport=5060;branch=z9hG4bK7551b3d6
From: "asterisk"sip:asterisk@202.141.151.35;tag=as382c9b91
To: sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws
Call-ID: 68ac7e7f38cbf60f4bc424c9167d1429@202.141.151.35:5060
CSeq: 102 OPTIONS
Content-Length: 0

SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=64239;received=202.141.151.100;branch=z9hG4bKqnSaIisdOMSHpOCLBpGd4koXFtz7WPB4
From: "milind"sip:1060@202.141.151.35;tag=uAd1ZHfTlZbvQyIzT8CT
To: "milind"sip:1060@202.141.151.35;tag=as6900d7f4
Contact: sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200
Call-ID: 76738299-e0fd-8ba5-df46-ef4a0cf6e0f8
CSeq: 7932 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.7.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 20 Mar 2014 12:23:02 GMT;20

SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=179:1
==session event = connected SIPml-api.js?svn=179:1
State machine: c0000_Started_2_Outgoing_X_oINVITE SIPml-api.js?svn=179:1
PeerConnectionClass = function RTCPeerConnection() { [native code] } SessionDescriptionClass = function RTCSessionDescription() { [native code] } IceCandidateClass = function RTCIceCandidate() { [native code] } SIPml-api.js?svn=179:1
ICE servers:[{“url”:“stun:stun.l.google.com:19302”}] SIPml-api.js?svn=179:1
==stack event = m_permission_requested SIPml-api.js?svn=179:1
==session event = connecting SIPml-api.js?svn=179:1
onGetUserMediaSuccess SIPml-api.js?svn=179:1
createOffer SIPml-api.js?svn=179:1
onCreateSdpSuccess SIPml-api.js?svn=179:1
==stack event = m_permission_accepted SIPml-api.js?svn=179:1
==session event = m_stream_audio_local_added SIPml-api.js?svn=179:1
onSetLocalDescriptionSuccess SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
ICE GATHERING COMPLETED! SIPml-api.js?svn=179:1
onIceGatheringCompleted SIPml-api.js?svn=179:1
SEND: INVITE sip:1@202.141.151.35 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKq2Gr2x4nHOJa7AdKTMlhXkhba1gKbwuU;rport
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35
Contact: "milind"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1060;ha1=d70c8dc5794b0e4cf735e0076b88bdec;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40315 INVITE
Content-Type: application/sdp
Content-Length: 2264
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.10
Organization: Doubango Telecom

v=0
o=- 1036746080042823600 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS SdisrjreH9vt3G7TeG5wrzIHX0ZEG8GQfIoP
m=audio 51902 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.106.1
a=rtcp:51902 IN IP4 192.168.106.1
a=candidate:4264415733 1 udp 2113937151 192.168.106.1 51902 typ host generation 0
a=candidate:4264415733 2 udp 2113937151 192.168.106.1 51902 typ host generation 0
a=candidate:4096368472 1 udp 2113937151 10.210.8.101 51903 typ host generation 0
a=candidate:4096368472 2 udp 2113937151 10.210.8.101 51903 typ host generation 0
a=candidate:1548745779 1 udp 2113937151 192.168.231.1 51904 typ host generation 0
a=candidate:1548745779 2 udp 2113937151 192.168.231.1 51904 typ host generation 0
a=candidate:2964204805 1 tcp 1509957375 192.168.106.1 0 typ host generation 0
a=candidate:2964204805 2 tcp 1509957375 192.168.106.1 0 typ host generation 0
a=candidate:3131728808 1 tcp 1509957375 10.210.8.101 0 typ host generation 0
a=candidate:3131728808 2 tcp 1509957375 10.210.8.101 0 typ host generation 0
a=candidate:315359427 1 tcp 1509957375 192.168.231.1 0 typ host generation 0
a=candidate:315359427 2 tcp 1509957375 192.168.231.1 0 typ host generation 0
a=ice-ufrag:xc/xlijFieGy0L7H
a=ice-pwd:4nT8rrBHk/G9r+d8X6yMLTeb
a=ice-options:google-ice
a=fingerprint:sha-256 89:CC:BF:E8:FB:F3:E3:4E:8C:A0:67:82:30:BD:80:78:E3:D8:A3:08:02:B5:C7:18:DD:9E:17:6C:C9:97:E7:21
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:h8Y7oemBZTfvBibZOio/EOLFl/8z9ipk7GuWnvDA
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CsJe7VgnurqRwJJVne1f9ASkRt4dKjSR2VJJonIR
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:4182659770 cname:PO+jD4lZ6N8p42MY
a=ssrc:4182659770 msid:SdisrjreH9vt3G7TeG5wrzIHX0ZEG8GQfIoP 7fb5aa92-9586-4b98-9f61-d19e5bbfa5bc
a=ssrc:4182659770 mslabel:SdisrjreH9vt3G7TeG5wrzIHX0ZEG8GQfIoP
a=ssrc:4182659770 label:7fb5aa92-9586-4b98-9f61-d19e5bbfa5bc
SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=64239;received=202.141.151.100;branch=z9hG4bKq2Gr2x4nHOJa7AdKTMlhXkhba1gKbwuU
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35;tag=as5cd8fb0c
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40315 INVITE
Content-Length: 0
Server: Asterisk PBX 11.7.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm=“202.141.151.35”,nonce=“58e689be”,stale=FALSE,algorithm=MD5

SIPml-api.js?svn=179:1
SEND: ACK sip:1@202.141.151.35 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKq2Gr2x4nHOJa7AdKTMlhXkhba1gKbwuU;rport
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35;tag=as5cd8fb0c
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40315 ACK
Content-Length: 0
Max-Forwards: 70

SIPml-api.js?svn=179:1
State machine: x0000_Any_2_Any_X_i401_407_INVITE SIPml-api.js?svn=179:1
SEND: INVITE sip:1@202.141.151.35 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKHu0ZBChtPGlGP3Oao5UnFJgbT5S1yWCb;rport
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35
Contact: "milind"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1060;ha1=d70c8dc5794b0e4cf735e0076b88bdec;+g.oma.sip-im;+sip.ice;language=“en,fr"
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40316 INVITE
Content-Type: application/sdp
Content-Length: 2264
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“202.141.151.35”,nonce=“58e689be”,uri="sip:1@202.141.151.35”,response=“d8f9eeaad6c0f30c818ab06a44a656c1”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.10
Organization: Doubango Telecom

v=0
o=- 1036746080042823600 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS SdisrjreH9vt3G7TeG5wrzIHX0ZEG8GQfIoP
m=audio 51902 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.106.1
a=rtcp:51902 IN IP4 192.168.106.1
a=candidate:4264415733 1 udp 2113937151 192.168.106.1 51902 typ host generation 0
a=candidate:4264415733 2 udp 2113937151 192.168.106.1 51902 typ host generation 0
a=candidate:4096368472 1 udp 2113937151 10.210.8.101 51903 typ host generation 0
a=candidate:4096368472 2 udp 2113937151 10.210.8.101 51903 typ host generation 0
a=candidate:1548745779 1 udp 2113937151 192.168.231.1 51904 typ host generation 0
a=candidate:1548745779 2 udp 2113937151 192.168.231.1 51904 typ host generation 0
a=candidate:2964204805 1 tcp 1509957375 192.168.106.1 0 typ host generation 0
a=candidate:2964204805 2 tcp 1509957375 192.168.106.1 0 typ host generation 0
a=candidate:3131728808 1 tcp 1509957375 10.210.8.101 0 typ host generation 0
a=candidate:3131728808 2 tcp 1509957375 10.210.8.101 0 typ host generation 0
a=candidate:315359427 1 tcp 1509957375 192.168.231.1 0 typ host generation 0
a=candidate:315359427 2 tcp 1509957375 192.168.231.1 0 typ host generation 0
a=ice-ufrag:xc/xlijFieGy0L7H
a=ice-pwd:4nT8rrBHk/G9r+d8X6yMLTeb
a=ice-options:google-ice
a=fingerprint:sha-256 89:CC:BF:E8:FB:F3:E3:4E:8C:A0:67:82:30:BD:80:78:E3:D8:A3:08:02:B5:C7:18:DD:9E:17:6C:C9:97:E7:21
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:h8Y7oemBZTfvBibZOio/EOLFl/8z9ipk7GuWnvDA
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:CsJe7VgnurqRwJJVne1f9ASkRt4dKjSR2VJJonIR
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
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:4182659770 cname:PO+jD4lZ6N8p42MY
a=ssrc:4182659770 msid:SdisrjreH9vt3G7TeG5wrzIHX0ZEG8GQfIoP 7fb5aa92-9586-4b98-9f61-d19e5bbfa5bc
a=ssrc:4182659770 mslabel:SdisrjreH9vt3G7TeG5wrzIHX0ZEG8GQfIoP
a=ssrc:4182659770 label:7fb5aa92-9586-4b98-9f61-d19e5bbfa5bc
SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=64239;received=202.141.151.100;branch=z9hG4bKHu0ZBChtPGlGP3Oao5UnFJgbT5S1yWCb
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35
Contact: sip:1@202.141.151.35:5060;transport=WS
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40316 INVITE
Content-Length: 0
Server: Asterisk PBX 11.7.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

SIPml-api.js?svn=179:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js?svn=179:1
==session event = i_ao_request SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=64239;received=202.141.151.100;branch=z9hG4bKHu0ZBChtPGlGP3Oao5UnFJgbT5S1yWCb
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35;tag=as31aa1936
Contact: sip:1@202.141.151.35:5060;transport=WS
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40316 INVITE
Content-Length: 0
Server: Asterisk PBX 11.7.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

SIPml-api.js?svn=179:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js?svn=179:1
==session event = i_ao_request SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=64239;received=202.141.151.100;branch=z9hG4bKHu0ZBChtPGlGP3Oao5UnFJgbT5S1yWCb
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35;tag=as31aa1936
Contact: sip:1@202.141.151.35:5060;transport=WS
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40316 INVITE
Content-Type: application/sdp
Content-Length: 577
Server: Asterisk PBX 11.7.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

v=0
o=root 822616138 822616138 IN IP4 202.141.151.35
s=Asterisk PBX 11.7.0
c=IN IP4 202.141.151.35
t=0 0
m=audio 13262 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:59911f931d39feb74d4b80103969a98c
a=ice-pwd:3a2f4f82063ade224c85cc3573d17cad
a=candidate:Hca8d9723 1 UDP 2130706431 202.141.151.35 13262 typ host
a=candidate:Hca8d9723 2 UDP 2130706430 202.141.151.35 13263 typ host
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:y6AXnHrQp8IGcGEFl0owbjL58wCmtSJw3yCoRBFX
SIPml-api.js?svn=179:1
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE SIPml-api.js?svn=179:1
setRemoteDescription(answer)
v=0
o=root 822616138 822616138 IN IP4 202.141.151.35
s=Asterisk PBX 11.7.0
c=IN IP4 202.141.151.35
t=0 0
m=audio 13262 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:59911f931d39feb74d4b80103969a98c
a=ice-pwd:3a2f4f82063ade224c85cc3573d17cad
a=candidate:Hca8d9723 1 UDP 2130706431 202.141.151.35 13262 typ host
a=candidate:Hca8d9723 2 UDP 2130706430 202.141.151.35 13263 typ host
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:y6AXnHrQp8IGcGEFl0owbjL58wCmtSJw3yCoRBFX
SIPml-api.js?svn=179:1
SEND: ACK sip:1@202.141.151.35:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKkExVS2C8H4wUUkw6yM6p;rport
From: "milind"sip:1060@202.141.151.35;tag=XTCQkIpxmvPhc4C8aT9C
To: sip:1@202.141.151.35;tag=as31aa1936
Contact: "milind"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 37797f75-15e9-375e-6802-2ac523c780f1
CSeq: 40316 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“202.141.151.35”,nonce=“58e689be”,uri=“sip:1@202.141.151.35:5060;transport=WS”,response=“bdc21eb5515386a5dfb18fabdbc8d359”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.03.10
Organization: Doubango Telecom

SIPml-api.js?svn=179:1
__on_add_stream SIPml-api.js?svn=179:1
onSetRemoteDescriptionSuccess SIPml-api.js?svn=179:1
==session event = m_early_media SIPml-api.js?svn=179:1
==session event = connected SIPml-api.js?svn=179:1
==session event = m_stream_audio_remote_added SIPml-api.js?svn=179:1
[/code] and also asterisk log is as follow== WebSocket connection from '202.141.151.100:64239' for protocol 'sip' accepted using version '13' -- Registered SIP '1060' at 202.141.151.100:64239 == Using SIP RTP CoS mark 5 -- Executing [1@default_1:1] Dial("SIP/1060-00000047", "SIP/milind,10") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/milind -- SIP/milind-00000048 is ringing -- SIP/milind-00000048 answered SIP/1060-00000047
plz help me to solve this problem…

navaismo · March 20, 2014, 1:04pm

If you installed webrtc2sip why are you connecting directly to asterisk?
If you are connecting directly to asterisk why are you using the breaker in the sipml5 config?

Show us the sip peer configuration.
Show us the asterisk sip debug.
Show us the asterisk rtp debug.

Based on your js log the sipml5 is using a local address in the sdp, so:
If your peer is in the same lan configure the Nat settings correctly in asterisk and use the null stun in sipml5.

milind_r · March 21, 2014, 7:17am

hi…thanx for your reply…i install doubango framework,webrtc2sip and asterisk on same machine i am attaching my sip debug info and sip configuration as belowÿINVITE sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP SIP/2.0 ÿVia: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK6c767db3;rport ÿMax-Forwards: 70 ÿFrom: "milind" <sip:1060@202.141.151.35>;tag=as53efef20 ÿTo: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> ÿContact: <sip:1060@202.141.151.35:5060> ÿCall-ID: 78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060 ÿCSeq: 102 INVITE ÿUser-Agent: Asterisk PBX 11.7.0 ÿDate: Fri, 21 Mar 2014 07:00:23 GMT ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ÿSupported: replaces, timer ÿContent-Type: application/sdp ÿContent-Length: 1315 ÿ ÿv=0 ÿo=root 1592107175 1592107175 IN IP4 202.141.151.35 ÿs=Asterisk PBX 11.7.0 ÿc=IN IP4 202.141.151.35 ÿt=0 0 ÿm=audio 13730 RTP/AVP 0 4 3 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101 ÿa=rtpmap:0 PCMU/8000 ÿa=rtpmap:4 G723/8000 ÿa=fmtp:4 annexa=no ÿa=rtpmap:3 GSM/8000 ÿa=rtpmap:8 PCMA/8000 ÿa=rtpmap:112 AAL2-G726-32/8000 ÿa=rtpmap:5 DVI4/8000 ÿa=rtpmap:7 LPC/8000 ÿa=rtpmap:18 G729/8000 ÿa=fmtp:18 annexb=no ÿa=rtpmap:110 speex/8000 ÿa=rtpmap:97 iLBC/8000 ÿa=fmtp:97 mode=30 ÿa=rtpmap:111 G726-32/8000 ÿa=rtpmap:9 G722/8000 ÿa=rtpmap:102 G7221/16000 ÿa=fmtp:102 bitrate=32000 ÿa=rtpmap:115 G7221/32000 ÿa=fmtp:115 bitrate=48000 ÿa=rtpmap:116 G719/48000 ÿa=fmtp:116 bitrate=64000 ÿa=rtpmap:117 speex/16000 ÿa=rtpmap:96 SILK/8000 ÿa=fmtp:96 maxaveragebitrate=10000 ÿa=fmtp:96 usedtx=0 ÿa=fmtp:96 useinbandfec=1 ÿa=rtpmap:100 SILK/12000 ÿa=fmtp:100 maxaveragebitrate=12000 ÿa=fmtp:100 usedtx=0 ÿa=fmtp:100 useinbandfec=1 ÿa=rtpmap:107 SILK/16000 ÿa=fmtp:107 maxaveragebitrate=20000 ÿa=fmtp:107 usedtx=0 ÿa=fmtp:107 useinbandfec=1 ÿa=rtpmap:108 SILK/24000 ÿa=fmtp:108 maxaveragebitrate=30000 ÿa=fmtp:108 usedtx=0 ÿa=fmtp:108 useinbandfec=1 ÿa=rtpmap:10 L16/8000 ÿa=rtpmap:118 L16/16000 ÿa=rtpmap:119 speex/32000 ÿa=rtpmap:101 telephone-event/8000 ÿa=fmtp:101 0-16 ÿa=ptime:20 ÿa=sendrecv ÿ ÿ--- [Mar 21 12:30:23] DEBUG[4746][C-0000003f] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15472 [Mar 21 12:30:23] DEBUG[4746][C-0000003f] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 202.141.151.100:59062 [Mar 21 12:30:23] VERBOSE[4746][C-0000003f] app_dial.c: -- Called SIP/kishor [Mar 21 12:30:23] VERBOSE[19001] chan_sip.c: ÿ<--- SIP read from UDP:202.141.151.100:59062 ---> ÿSIP/2.0 180 Ringing ÿVia: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK6c767db3;rport=5060 ÿContact: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> ÿTo: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP>;tag=47c98876 ÿFrom: "milind"<sip:1060@202.141.151.35>;tag=as53efef20 ÿCall-ID: 78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060 ÿCSeq: 102 INVITE ÿUser-Agent: Z 3.2.21357 r21103 ÿContent-Length: 0 ÿ ÿ<-------------> [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK6c767db3;rport=5060 [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 2 [ 81]: Contact: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 3 [ 89]: To: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP>;tag=47c98876 [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 4 [ 54]: From: "milind"<sip:1060@202.141.151.35>;tag=as53efef20 [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 5 [ 61]: Call-ID: 78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060 [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 7 [ 30]: User-Agent: Z 3.2.21357 r21103 [Mar 21 12:30:23] DEBUG[19001] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Mar 21 12:30:23] VERBOSE[19001] chan_sip.c: --- (9 headers 0 lines) --- [Mar 21 12:30:23] DEBUG[19001][C-0000003f] chan_sip.c: *** SIP TIMER: Cancelling retransmission #15472 - INVITE (got response) [Mar 21 12:30:23] DEBUG[19001][C-0000003f] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060' Request 102: Found [Mar 21 12:30:23] DEBUG[19001][C-0000003f] chan_sip.c: SIP response 180 to standard invite [Mar 21 12:30:23] DEBUG[19001][C-0000003f] chan_sip.c: build_route: Contact hop: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> [Mar 21 12:30:23] VERBOSE[19001][C-0000003f] chan_sip.c: list_route: hop: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> [Mar 21 12:30:23] DEBUG[18986] devicestate.c: No provider found, checking channel drivers for SIP - kishor [Mar 21 12:30:23] DEBUG[18986] chan_sip.c: Checking device state for peer kishor [Mar 21 12:30:23] DEBUG[18986] devicestate.c: Changing state for SIP/kishor - state 1 (Not in use) [Mar 21 12:30:23] DEBUG[18986] devicestate.c: device 'SIP/kishor' state '1' [Mar 21 12:30:23] DEBUG[19024] app_queue.c: Device 'SIP/kishor' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 12:30:23] VERBOSE[4746][C-0000003f] app_dial.c: -- SIP/kishor-00000066 is ringing [Mar 21 12:30:23] DEBUG[4746][C-0000003f] rtp_engine.c: Setting early bridge SDP of 'SIP/1060-00000065' with that of 'SIP/kishor-00000066' [Mar 21 12:30:23] DEBUG[4746][C-0000003f] chan_sip.c: Trying to put 'SIP/2.0 180' onto WS socket destined for 202.141.151.100:52908 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Allocating new SIP dialog for 043232fa67b64e374c34aea77d177719@202.141.151.35:5060 - OPTIONS (No RTP) [Mar 21 12:30:24] DEBUG[19001] acl.c: For destination '202.141.151.100', our source address is '202.141.151.35'. [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 202.141.151.35:5060 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: SIP call-id changed from '043232fa67b64e374c34aea77d177719@202.141.151.35:5060' to '7e99274b0257715c5e7747474cbfb2cb@202.141.151.35:5060' [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Initializing initreq for method OPTIONS - callid 7e99274b0257715c5e7747474cbfb2cb@202.141.151.35:5060 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 0 [ 86]: OPTIONS sip:milind@10.210.8.101:57129;rinstance=fb1b76b9b58ce398;transport=UDP SIP/2.0 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK6130253f;rport [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 3 [ 61]: From: "asterisk" <sip:asterisk@202.141.151.35>;tag=as32b1d96d [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 4 [ 76]: To: <sip:milind@10.210.8.101:57129;rinstance=fb1b76b9b58ce398;transport=UDP> [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 5 [ 43]: Contact: <sip:asterisk@202.141.151.35:5060> [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 6 [ 61]: Call-ID: 7e99274b0257715c5e7747474cbfb2cb@202.141.151.35:5060 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 11.7.0 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 9 [ 35]: Date: Fri, 21 Mar 2014 07:00:24 GMT [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15475 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 202.141.151.100:57129 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK6130253f;rport=5060 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 2 [ 33]: Contact: <sip:10.210.8.101:57129> [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 3 [ 89]: To: <sip:milind@10.210.8.101:57129;rinstance=fb1b76b9b58ce398;transport=UDP>;tag=e432ba0e [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 4 [ 60]: From: "asterisk"<sip:asterisk@202.141.151.35>;tag=as32b1d96d [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 5 [ 61]: Call-ID: 7e99274b0257715c5e7747474cbfb2cb@202.141.151.35:5060 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 7 [ 40]: Accept: application/sdp, application/sdp [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 10 [ 74]: Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 11 [ 30]: User-Agent: Z 3.2.21357 r21367 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 12 [ 28]: Allow-Events: presence, kpml [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15475 [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Stopping retransmission on '7e99274b0257715c5e7747474cbfb2cb@202.141.151.35:5060' of Request 102: Match Found [Mar 21 12:30:24] NOTICE[19001] chan_sip.c: Peer 'milind' is now Reachable. (4ms / 2000ms) [Mar 21 12:30:24] DEBUG[19001] chan_sip.c: Destroying SIP dialog 7e99274b0257715c5e7747474cbfb2cb@202.141.151.35:5060 [Mar 21 12:30:24] DEBUG[18986] devicestate.c: No provider found, checking channel drivers for SIP - milind [Mar 21 12:30:24] DEBUG[18986] chan_sip.c: Checking device state for peer milind [Mar 21 12:30:24] DEBUG[18986] devicestate.c: Changing state for SIP/milind - state 1 (Not in use) [Mar 21 12:30:24] DEBUG[18986] devicestate.c: device 'SIP/milind' state '1' [Mar 21 12:30:24] DEBUG[19024] app_queue.c: Device 'SIP/milind' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 0 [ 0]: [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Allocating new SIP dialog for 37fac1d32ba93f676749777f7ecfa450@202.141.151.35:5060 - OPTIONS (No RTP) [Mar 21 12:30:26] DEBUG[19001] acl.c: For destination '202.141.151.100', our source address is '202.141.151.35'. [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 202.141.151.35:5060 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: SIP call-id changed from '37fac1d32ba93f676749777f7ecfa450@202.141.151.35:5060' to '4f41cd2767a38e7362f011840dd6ffde@202.141.151.35:5060' [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Initializing initreq for method OPTIONS - callid 4f41cd2767a38e7362f011840dd6ffde@202.141.151.35:5060 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 0 [ 86]: OPTIONS sip:mahesh@10.210.8.110:53215;rinstance=c17b86d57e741912;transport=UDP SIP/2.0 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK0e75fbaa;rport [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 3 [ 61]: From: "asterisk" <sip:asterisk@202.141.151.35>;tag=as0751eb30 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 4 [ 76]: To: <sip:mahesh@10.210.8.110:53215;rinstance=c17b86d57e741912;transport=UDP> [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 5 [ 43]: Contact: <sip:asterisk@202.141.151.35:5060> [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 6 [ 61]: Call-ID: 4f41cd2767a38e7362f011840dd6ffde@202.141.151.35:5060 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 11.7.0 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 9 [ 35]: Date: Fri, 21 Mar 2014 07:00:26 GMT [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15478 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 202.141.151.100:53215 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK0e75fbaa;rport=5060 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 2 [ 33]: Contact: <sip:10.210.8.110:53215> [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 3 [ 89]: To: <sip:mahesh@10.210.8.110:53215;rinstance=c17b86d57e741912;transport=UDP>;tag=0310cf20 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 4 [ 60]: From: "asterisk"<sip:asterisk@202.141.151.35>;tag=as0751eb30 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 5 [ 61]: Call-ID: 4f41cd2767a38e7362f011840dd6ffde@202.141.151.35:5060 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 7 [ 40]: Accept: application/sdp, application/sdp [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 10 [ 74]: Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 11 [ 30]: User-Agent: Z 3.1.21132 r21103 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 12 [ 28]: Allow-Events: presence, kpml [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Header 13 [ 17]: Content-Length: 0 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15478 [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Stopping retransmission on '4f41cd2767a38e7362f011840dd6ffde@202.141.151.35:5060' of Request 102: Match Found [Mar 21 12:30:26] DEBUG[19001] chan_sip.c: Destroying SIP dialog 4f41cd2767a38e7362f011840dd6ffde@202.141.151.35:5060 [Mar 21 12:30:29] DEBUG[4746][C-0000003f] res_rtp_asterisk.c: 0x7fc5980582a0 -- Probation learning mode pass with source address 202.141.151.100:8000 [Mar 21 12:30:29] DEBUG[4746][C-0000003f] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fc598053d58' [Mar 21 12:30:29] DEBUG[4746][C-0000003f] res_rtp_asterisk.c: 0x7fc5980582a0 -- Probation learning mode pass with source address 202.141.151.100:8000 [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: ÿ<--- SIP read from UDP:202.141.151.100:59062 ---> ÿSIP/2.0 200 OK ÿVia: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK6c767db3;rport=5060 ÿContact: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> ÿTo: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP>;tag=47c98876 ÿFrom: "milind"<sip:1060@202.141.151.35>;tag=as53efef20 ÿCall-ID: 78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060 ÿCSeq: 102 INVITE ÿAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE ÿContent-Type: application/sdp ÿSupported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri ÿUser-Agent: Z 3.2.21357 r21103 ÿAllow-Events: presence, kpml ÿContent-Length: 239 ÿ ÿv=0 ÿo=Z 0 2 IN IP4 10.210.9.156 ÿs=Z ÿc=IN IP4 10.210.9.156 ÿt=0 0 ÿm=audio 8000 RTP/AVP 0 3 110 8 98 101 ÿa=rtpmap:110 speex/8000 ÿa=rtpmap:98 iLBC/8000 ÿa=fmtp:98 mode=20 ÿa=rtpmap:101 telephone-event/8000 ÿa=fmtp:101 0-15 ÿa=sendrecv ÿ<-------------> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK6c767db3;rport=5060 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 2 [ 81]: Contact: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 3 [ 89]: To: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP>;tag=47c98876 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 4 [ 54]: From: "milind"<sip:1060@202.141.151.35>;tag=as53efef20 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 5 [ 61]: Call-ID: 78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 9 [ 74]: Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 10 [ 30]: User-Agent: Z 3.2.21357 r21103 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 11 [ 28]: Allow-Events: presence, kpml [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 12 [ 19]: Content-Length: 239 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 13 [ 0]: [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 0 [ 3]: v=0 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 1 [ 27]: o=Z 0 2 IN IP4 10.210.9.156 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 2 [ 3]: s=Z [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 3 [ 21]: c=IN IP4 10.210.9.156 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 4 [ 5]: t=0 0 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 5 [ 37]: m=audio 8000 RTP/AVP 0 3 110 8 98 101 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 6 [ 23]: a=rtpmap:110 speex/8000 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 7 [ 21]: a=rtpmap:98 iLBC/8000 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 8 [ 17]: a=fmtp:98 mode=20 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 11 [ 10]: a=sendrecv [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: --- (13 headers 12 lines) --- [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Acked pending invite 102 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Stopping retransmission on '78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060' of Request 102: Match Found [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: SIP response 200 to standard invite [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing session-level SDP o=Z 0 2 IN IP4 10.210.9.156... OK. [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing session-level SDP s=Z... UNSUPPORTED OR FAILED. [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing session-level SDP c=IN IP4 10.210.9.156... OK. [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found RTP audio format 0 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Setting payload 0 based on m type on 0x7fc5aaa1c3d0 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found RTP audio format 3 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Setting payload 3 based on m type on 0x7fc5aaa1c3d0 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found RTP audio format 110 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Setting payload 110 based on m type on 0x7fc5aaa1c3d0 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found RTP audio format 8 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Setting payload 8 based on m type on 0x7fc5aaa1c3d0 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found RTP audio format 98 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Setting payload 98 based on m type on 0x7fc5aaa1c3d0 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found RTP audio format 101 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Setting payload 101 based on m type on 0x7fc5aaa1c3d0 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found audio description format speex for ID 110 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000... OK. [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found audio description format iLBC for ID 98 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 iLBC/8000... OK. [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing media-level (audio) SDP a=fmtp:98 mode=20... OK. [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Found audio description format telephone-event for ID 101 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|speex|ilbc) [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Mar 21 12:30:29] DEBUG[19001][C-0000003f] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fc598053d58' [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Peer audio RTP is at port 10.210.9.156:8000 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Copying payload 0 from 0x7fc5aaa1c3d0 to 0x7fc598053f20 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Copying payload 3 from 0x7fc5aaa1c3d0 to 0x7fc598053f20 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Copying payload 8 from 0x7fc5aaa1c3d0 to 0x7fc598053f20 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Copying payload 98 from 0x7fc5aaa1c3d0 to 0x7fc598053f20 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Copying payload 101 from 0x7fc5aaa1c3d0 to 0x7fc598053f20 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] rtp_engine.c: Copying payload 110 from 0x7fc5aaa1c3d0 to 0x7fc598053f20 [Mar 21 12:30:29] DEBUG[19001][C-0000003f] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fc598053d58' [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: We're settling with these formats: (gsm|ulaw|alaw|speex|ilbc) [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Setting native formats after processing SDP. peer joint formats (gsm|ulaw|alaw|speex|ilbc), old nativeformats (ulaw) [Mar 21 12:30:29] DEBUG[19001][C-0000003f] format_pref.c: Could not find preferred codec - Going for the best codec [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Updating call counter for outgoing call [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: build_route: Contact hop: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: list_route: hop: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Strict routing enforced for session 78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: set_destination: Parsing <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP> for address/port to send to [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: set_destination: set destination to 10.210.9.156:59062 [Mar 21 12:30:29] VERBOSE[19001][C-0000003f] chan_sip.c: Transmitting (NAT) to 202.141.151.100:59062: ÿACK sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP SIP/2.0 ÿVia: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK22c5e3d8;rport ÿMax-Forwards: 70 ÿFrom: "milind" <sip:1060@202.141.151.35>;tag=as53efef20 ÿTo: <sip:kishor@10.210.9.156:59062;rinstance=4d403be451fd0ef8;transport=UDP>;tag=47c98876 ÿContact: <sip:1060@202.141.151.35:5060> ÿCall-ID: 78b5a82a7442b28528d5e94143a6a352@202.141.151.35:5060 ÿCSeq: 102 ACK ÿUser-Agent: Asterisk PBX 11.7.0 ÿContent-Length: 0 ÿ ÿ ÿ--- [Mar 21 12:30:29] DEBUG[19001][C-0000003f] chan_sip.c: Trying to put 'ACK sip:kis' onto UDP socket destined for 202.141.151.100:59062 [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: ÿ<--- SIP read from UDP:202.141.151.100:59062 ---> ÿPUBLISH sip:kishor@202.141.151.35;transport=UDP SIP/2.0 ÿVia: SIP/2.0/UDP 10.210.9.156:59062;branch=z9hG4bK-d8754z-de57d4d575ea85c9-1---d8754z- ÿMax-Forwards: 70 ÿContact: <sip:kishor@10.210.9.156:59062;transport=UDP> ÿTo: <sip:kishor@202.141.151.35;transport=UDP> ÿFrom: <sip:kishor@202.141.151.35;transport=UDP>;tag=70a27569 ÿCall-ID: NjI4Y2IwNzJlZTA0NGI0MjgwOTcxZTk0NDBjNjU1MWQ. ÿCSeq: 1 PUBLISH ÿExpires: 600 ÿAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE ÿContent-Type: application/pidf+xml ÿSupported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri ÿUser-Agent: Z 3.2.21357 r21103 ÿEvent: presence ÿAllow-Events: presence, kpml ÿContent-Length: 272 ÿ ÿ<?xml version="1.0" encoding="UTF-8"?> ÿ<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:kishor@202.141.151.35;transport=UDP"> <tuple id="kishor" > <status><basic>open</basic></status> <note>On the phone</note> </tuple> ÿ</presence> ÿ<-------------> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 0 [ 55]: PUBLISH sip:kishor@202.141.151.35;transport=UDP SIP/2.0 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 1 [ 86]: Via: SIP/2.0/UDP 10.210.9.156:59062;branch=z9hG4bK-d8754z-de57d4d575ea85c9-1---d8754z- [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 3 [ 54]: Contact: <sip:kishor@10.210.9.156:59062;transport=UDP> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 4 [ 45]: To: <sip:kishor@202.141.151.35;transport=UDP> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 5 [ 60]: From: <sip:kishor@202.141.151.35;transport=UDP>;tag=70a27569 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 6 [ 53]: Call-ID: NjI4Y2IwNzJlZTA0NGI0MjgwOTcxZTk0NDBjNjU1MWQ. [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 7 [ 15]: CSeq: 1 PUBLISH [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 8 [ 12]: Expires: 600 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 10 [ 34]: Content-Type: application/pidf+xml [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 11 [ 74]: Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 12 [ 30]: User-Agent: Z 3.2.21357 r21103 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 13 [ 15]: Event: presence [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 14 [ 28]: Allow-Events: presence, kpml [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 15 [ 19]: Content-Length: 272 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Header 16 [ 0]: [Mar 21 12:30:29] VERBOSE[4746][C-0000003f] app_dial.c: -- SIP/kishor-00000066 answered SIP/1060-00000065 [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 0 [ 38]: <?xml version="1.0" encoding="UTF-8"?> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 1 [188]: <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:kishor@202.141.151.35;transport=UDP"> <tuple id="kishor" > <status><basic>open</basic></status> <note>On the phone</note> </tuple> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Body 2 [ 11]: </presence> [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: --- (16 headers 3 lines) --- [Mar 21 12:30:29] DEBUG[4746][C-0000003f] rtp_engine.c: Setting early bridge SDP of 'SIP/1060-00000065' with that of 'SIP/kishor-00000066' [Mar 21 12:30:29] DEBUG[18986] devicestate.c: No provider found, checking channel drivers for SIP - kishor [Mar 21 12:30:29] DEBUG[4746][C-0000003f] chan_sip.c: SIP answering channel: SIP/1060-00000065 [Mar 21 12:30:29] DEBUG[19001] acl.c: For destination '202.141.151.100', our source address is '202.141.151.35'. [Mar 21 12:30:29] DEBUG[4746][C-0000003f] res_rtp_asterisk.c: Setting the marker bit due to a source update [Mar 21 12:30:29] DEBUG[18986] chan_sip.c: Checking device state for peer kishor [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 202.141.151.35:5060 [Mar 21 12:30:29] DEBUG[18986] devicestate.c: Changing state for SIP/kishor - state 1 (Not in use) [Mar 21 12:30:29] DEBUG[18986] devicestate.c: device 'SIP/kishor' state '1' [Mar 21 12:30:29] DEBUG[18986] devicestate.c: No provider found, checking channel drivers for SIP - 1060 [Mar 21 12:30:29] DEBUG[18986] chan_sip.c: Checking device state for peer 1060 [Mar 21 12:30:29] DEBUG[18986] devicestate.c: Changing state for SIP/1060 - state 1 (Not in use) [Mar 21 12:30:29] DEBUG[18986] devicestate.c: device 'SIP/1060' state '1' [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: NAT detected for 10.210.9.156 / 202.141.151.100 [Mar 21 12:30:29] DEBUG[4746][C-0000003f] chan_sip.c: Setting framing from config on incoming call [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: Sending to 202.141.151.100:59062 (NAT) [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Allocating new SIP dialog for NjI4Y2IwNzJlZTA0NGI0MjgwOTcxZTk0NDBjNjU1MWQ. - PUBLISH (No RTP) [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: **** Received PUBLISH (15) - Command in SIP PUBLISH [Mar 21 12:30:29] DEBUG[19024] app_queue.c: Device 'SIP/kishor' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 12:30:29] DEBUG[19024] app_queue.c: Device 'SIP/1060' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 12:30:29] DEBUG[4746][C-0000003f] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True [Mar 21 12:30:29] DEBUG[4746][C-0000003f] chan_sip.c: ** Our prefcodec: (nothing) [Mar 21 12:30:29] DEBUG[4746][C-0000003f] sip/sdp_crypto.c: Crypto line: a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:Taqqjbwtj0p0SayB+YYLNNw2tVOdXE2x1/77aOmz [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: ÿ<--- Transmitting (NAT) to 202.141.151.100:59062 ---> ÿSIP/2.0 489 Bad Event ÿVia: SIP/2.0/UDP 10.210.9.156:59062;branch=z9hG4bK-d8754z-de57d4d575ea85c9-1---d8754z-;received=202.141.151.100;rport=59062 ÿFrom: <sip:kishor@202.141.151.35;transport=UDP>;tag=70a27569 ÿTo: <sip:kishor@202.141.151.35;transport=UDP>;tag=as7614730f ÿCall-ID: NjI4Y2IwNzJlZTA0NGI0MjgwOTcxZTk0NDBjNjU1MWQ. ÿCSeq: 1 PUBLISH ÿServer: Asterisk PBX 11.7.0 ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH ÿSupported: replaces, timer ÿContent-Length: 0 ÿ ÿ ÿ<------------> [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Trying to put 'SIP/2.0 489' onto UDP socket destined for 202.141.151.100:59062 [Mar 21 12:30:29] DEBUG[4746][C-0000003f] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: SIP message could not be handled, bad request: NjI4Y2IwNzJlZTA0NGI0MjgwOTcxZTk0NDBjNjU1MWQ. [Mar 21 12:30:29] DEBUG[19001] chan_sip.c: Destroying SIP dialog NjI4Y2IwNzJlZTA0NGI0MjgwOTcxZTk0NDBjNjU1MWQ. [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: Really destroying SIP dialog 'NjI4Y2IwNzJlZTA0NGI0MjgwOTcxZTk0NDBjNjU1MWQ.' Method: PUBLISH [Mar 21 12:30:29] DEBUG[4746][C-0000003f] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw) [Mar 21 12:30:29] DEBUG[4746][C-0000003f] chan_sip.c: Trying to put 'SIP/2.0 200' onto WS socket destined for 202.141.151.100:52908 [Mar 21 12:30:29] VERBOSE[19001] chan_sip.c: ÿ<--- SIP read from UDP:202.141.151.100:59062 ---> ÿSUBSCRIBE sip:kishor@202.141.151.35;transport=UDP SIP/2.0 ÿVia: SIP/2.0/UDP 10.210.9.156:59062;branch=z9hG4bK-d8754z-af41a6d5d0baa9d8-1---d8754z- ÿMax-Forwards: 70 ÿContact: <sip:kishor@10.210.9.156:59062;transport=UDP> ÿTo: <sip:kishor@202.141.151.35;transport=UDP> ÿFrom: <sip:kishor@202.141.151.35;transport=UDP>;tag=2878f77e ÿCall-ID: ZDdkOTUzODZjODFmYjc3Y2NlMmRiOTg4MTljNDliODk. ÿCSeq: 1 SUBSCRIBE ÿExpires: 600 ÿAccept: application/watcherinfo+xml ÿAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE ÿSupported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri ÿUser-Agent: Z 3.2.21357 r21103 ÿEvent: presence.winfo ÿAllow-Events: presence, kpml ÿContent-Length: 0

navaismo · March 21, 2014, 1:10pm

[quote]If you installed webrtc2sip why are you connecting directly to asterisk?
If you are connecting directly to asterisk why are you using the breaker in the sipml5 config?

Show us the sip peer configuration.
Show us the asterisk sip debug.
Show us the asterisk rtp debug.[/quote]

Show us the mentioned debugs separated and without debug console information that make unreadable the info. And first fix your configuration in the sipml5 settings.

milind_r · March 21, 2014, 4:43pm

hi…thank you for your reply…i am not using webrtc2sip so i disable the webrtc breaker in expert mode. i attached here required file for debug
sipml js consoleSIPML5 API version = 1.3.214 SIPml-api.js?svn=179:1 location=http://sipml5.org/call.htm?svn=203# call.htm?svn=203:147 event.returnValue is deprecated. Please use the standard event.preventDefault() instead. jquery.js:2 2 Not implemented SIPml-api.js?svn=179:1 tsk_utils_log_error SIPml-api.js?svn=179:1 tsip_dialog_layer.handle_incoming_message SIPml-api.js?svn=179:3 tsip_transport_layer.handle_incoming_message SIPml-api.js?svn=179:3 __tsip_transport_ws_onmessage SIPml-api.js?svn=179:3
sip debug console copy from asterisk cli after connection is established between SIMPL5 and softphone[code]<?xml version="1.0" encoding="UTF-8"?>
open On the phone

<------------->
— (16 headers 3 lines) —
Sending to 202.141.151.100:59062 (NAT)

<— Transmitting (NAT) to 202.141.151.100:59062 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.210.9.156:59062;branch=z9hG4bK-d8754z-093661a2b8c01f9e-1—d8754z-;received=202.141.151.100;rport=59062
From: sip:kishor@202.141.151.35;transport=UDP;tag=19b6b468
To: sip:kishor@202.141.151.35;transport=UDP;tag=as3485855f
Call-ID: NmE0MjE1NGZiYmU2ZjMxNTU0YTVhNmNmY2Y3MDZmMjc.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘NmE0MjE1NGZiYmU2ZjMxNTU0YTVhNmNmY2Y3MDZmMjc.’ Method: PUBLISH
cmj-nsdg-svr1*CLI> core set debug 1
Core debug was 0 and is now 1

<— SIP read from UDP:202.141.151.100:59062 —>

<------------->
Reliably Transmitting (NAT) to 202.141.151.100:59062:
OPTIONS sip:kishor@10.210.9.156:59062;rinstance=6ae3a53fc1a7e383;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK0b817619;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@202.141.151.35;tag=as55a1f94f
To: sip:kishor@10.210.9.156:59062;rinstance=6ae3a53fc1a7e383;transport=UDP
Contact: sip:asterisk@202.141.151.35:5060
Call-ID: 791c94493496b7c934b6e2de519ba3b3@202.141.151.35:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0
Date: Fri, 21 Mar 2014 16:36:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:202.141.151.100:59062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.141.151.35:5060;branch=z9hG4bK0b817619;rport=5060
Contact: sip:10.210.9.156:59062
To: sip:kishor@10.210.9.156:59062;rinstance=6ae3a53fc1a7e383;transport=UDP;tag=b211f656
From: "asterisk"sip:asterisk@202.141.151.35;tag=as55a1f94f
Call-ID: 791c94493496b7c934b6e2de519ba3b3@202.141.151.35:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21103
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘791c94493496b7c934b6e2de519ba3b3@202.141.151.35:5060’ Method: OPTIONS
[/code]

and my sip configuration[code];[global]
;autoframing=yes
[general]
udpbindaddr=0.0.0.0:5060
realm=202.141.151.35
transport=udp,ws

[8000]
secret=8000
;context=demo
context=default_1
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,ws
encryption=yes
dial=SIP/8000
;callerid=Sanjay Willie <8001>
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no

[kishor]
qualify=yes
username=kishor
secret=Welcome123
host=dynamic
callerid=123456
type=friend
context=default_1
;context=festival_test
;context=demo
;context=from_pstn
Disallow=all
Allow=all
;Allow=gsm
;Allow=ulaw
;Allow=alaw
nat=yes
[/code]

navaismo · March 21, 2014, 4:50pm

Your debugs are incomplete.

Also check that icesupport is enabled in the rtp.conf and tell us if the sipml5 client is in the same network and finally attach a part of the asterisk rtp debug to see the incoming and outgoing addresses

milind_r · March 24, 2014, 1:57pm

hi…navaismo…thankx for your reply… i checked my rtp.conf and it has icesupport=yes. my asterisk server running on public ip address 202.141.151.35 and now my webclient ip address is 10.210.8.110 which in private network…and i am calling to the client now which is on private ip address 10.210.8.114… i am attaching the rtp debug for debugging perpose. for the call set as follow

webclient(10.210.8.110–private Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from Sent RTP packet to Got RTP packet from ip)<======>asterisk(202.141.151.35–public ip)<=======>sip client(10.210.8.114 --private ip). but i am not understand the roll of the ip address 202.141.151.100…where does the host on this ip address come in picture?..plz reply…Got RTP packet from 202.141.151.100:8000 (type 00, seq 008852, ts 499546764, len 000160) 10.210.8.110:54247 (type 00, seq 033005, ts 499546760, len 000164) 202.141.151.100:8000 (type 00, seq 008853, ts 499546924, len 000160) 10.210.8.110:54247 (type 00, seq 033006, ts 499546920, len 000164) 202.141.151.100:8000 (type 00, seq 008854, ts 499547084, len 000160) 10.210.8.110:54247 (type 00, seq 033007, ts 499547080, len 000164) 202.141.151.100:8000 (type 00, seq 008855, ts 499547244, len 000160) 10.210.8.110:54247 (type 00, seq 033008, ts 499547240, len 000164) 202.141.151.100:8000 (type 00, seq 008856, ts 499547404, len 000160) 10.210.8.110:54247 (type 00, seq 033009, ts 499547400, len 000164) 202.141.151.100:8000 (type 00, seq 008857, ts 499547564, len 000160) 10.210.8.110:54247 (type 00, seq 033010, ts 499547560, len 000164) 202.141.151.100:8000 (type 00, seq 008858, ts 499547724, len 000160) 10.210.8.110:54247 (type 00, seq 033011, ts 499547720, len 000164) 202.141.151.100:8000 (type 00, seq 008859, ts 499547884, len 000160) 10.210.8.110:54247 (type 00, seq 033012, ts 499547880, len 000164) 202.141.151.100:8000 (type 00, seq 008860, ts 499548044, len 000160) 10.210.8.110:54247 (type 00, seq 033013, ts 499548040, len 000164) 202.141.151.100:8000 (type 00, seq 008861, ts 499548204, len 000160) 10.210.8.110:54247 (type 00, seq 033014, ts 499548200, len 000164) 202.141.151.100:8000 (type 00, seq 008862, ts 499548364, len 000160) 10.210.8.110:54247 (type 00, seq 033015, ts 499548360, len 000164) 202.141.151.100:8000 (type 00, seq 008863, ts 499548524, len 000160) 10.210.8.110:54247 (type 00, seq 033016, ts 499548520, len 000164) 202.141.151.100:8000 (type 00, seq 008864, ts 499548684, len 000160) 10.210.8.110:54247 (type 00, seq 033017, ts 499548680, len 000164) 202.141.151.100:8000 (type 00, seq 008865, ts 499548844, len 000160) 10.210.8.110:54247 (type 00, seq 033018, ts 499548840, len 000164) 202.141.151.100:8000 (type 00, seq 008866, ts 499549004, len 000160) 10.210.8.110:54247 (type 00, seq 033019, ts 499549000, len 000164) 202.141.151.100:8000 (type 00, seq 008867, ts 499549164, len 000160) 10.210.8.110:54247 (type 00, seq 033020, ts 499549160, len 000164) 202.141.151.100:8000 (type 00, seq 008868, ts 499549324, len 000160) 10.210.8.110:54247 (type 00, seq 033021, ts 499549320, len 000164) 202.141.151.100:8000 (type 00, seq 008869, ts 499549484, len 000160) 10.210.8.110:54247 (type 00, seq 033022, ts 499549480, len 000164) 202.141.151.100:8000 (type 00, seq 008870, ts 499549644, len 000160) 10.210.8.110:54247 (type 00, seq 033023, ts 499549640, len 000164) 202.141.151.100:8000 (type 00, seq 008871, ts 499549804, len 000160) 10.210.8.110:54247 (type 00, seq 033024, ts 499549800, len 000164) 202.141.151.100:8000 (type 00, seq 008872, ts 499549964, len 000160) 10.210.8.110:54247 (type 00, seq 033025, ts 499549960, len 000164) 202.141.151.100:8000 (type 00, seq 008873, ts 499550124, len 000160) 10.210.8.110:54247 (type 00, seq 033026, ts 499550120, len 000164) 202.141.151.100:8000 (type 00, seq 008874, ts 499550284, len 000160) 10.210.8.110:54247 (type 00, seq 033027, ts 499550280, len 000164) 202.141.151.100:8000 (type 00, seq 008875, ts 499550444, len 000160) 10.210.8.110:54247 (type 00, seq 033028, ts 499550440, len 000164) 202.141.151.100:8000 (type 00, seq 008876, ts 499550604, len 000160) 10.210.8.110:54247 (type 00, seq 033029, ts 499550600, len 000164) 202.141.151.100:8000 (type 00, seq 008877, ts 499550764, len 000160) 10.210.8.110:54247 (type 00, seq 033030, ts 499550760, len 000164) 202.141.151.100:8000 (type 00, seq 008878, ts 499550924, len 000160) 10.210.8.110:54247 (type 00, seq 033031, ts 499550920, len 000164) 202.141.151.100:8000 (type 00, seq 008879, ts 499551084, len 000160) 10.210.8.110:54247 (type 00, seq 033032, ts 499551080, len 000164) 202.141.151.100:8000 (type 00, seq 008880, ts 499551244, len 000160) 10.210.8.110:54247 (type 00, seq 033033, ts 499551240, len 000164) 202.141.151.100:8000 (type 00, seq 008881, ts 499551404, len 000160) 10.210.8.110:54247 (type 00, seq 033034, ts 499551400, len 000164) 202.141.151.100:8000 (type 00, seq 008882, ts 499551564, len 000160) 10.210.8.110:54247 (type 00, seq 033035, ts 499551560, len 000164) 202.141.151.100:8000 (type 00, seq 008883, ts 499551724, len 000160) 10.210.8.110:54247 (type 00, seq 033036, ts 499551720, len 000164) 202.141.151.100:8000 (type 00, seq 008884, ts 499551884, len 000160) 10.210.8.110:54247 (type 00, seq 033037, ts 499551880, len 000164) 202.141.151.100:8000 (type 00, seq 008885, ts 499552044, len 000160) 10.210.8.110:54247 (type 00, seq 033038, ts 499552040, len 000164) 202.141.151.100:8000 (type 00, seq 008886, ts 499552204, len 000160)

navaismo · March 24, 2014, 5:03pm

Two things:

–If your asterisk is in a Public IP then your peers must use NAT=yes and send the Public address of their location, so based in your RTP debug you didn’t configured asterisk and the peers correctly.

–Your RTP debug show local ip address thats wrong for a public asterisk setup and also the RTP didn’t show the VIA ICE string.

Check this thread–>http://forums.digium.com/viewtopic.php?f=1&t=88761&p=194687&hilit=via+ice&sid=6895e6f701b7f091af41a536ef86e3c7#p194687

david55 · March 24, 2014, 6:46pm

nat=yes is a deprecated option. It often isn’t necessary.

milind_r · April 21, 2014, 11:16am

yes…it is acually the NAT issue… i got the solution thank you very much for your guide.