rozasmerti / sipml5

Automatically exported from code.google.com/p/sipml5
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Call established but no audio on both ends #132

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago
Problem:  (SIP) call setup successful. There was ringing on both ends. But no 
audio on both ends.

Setup:
Google-Chrome   ---(WS/WSS)------> Asterisk Server -------->SIP Phone
All under a local network, with internet.

sipml5 :Version 1.3.203
chrome : Version 30.0.1599.101
Asterisk: 11.5

Same issue for SIP over WS or WSS.

Other Observation: Wireshark dumps on client laptop and the Asterisk-server 
show clearly, only one-way audio (RTP / SRTP traffic) from SIP Phone  ---> 
Asterisk  ----> Chrome.

hence, 2 sub-issues in chrome
1. Received SRTP packets, not processed in Chrome
2. No microphone-input aaudio, sent out.

What steps will reproduce the problem?
1.From chrome(V30), do a SIP register to Asterisk(11.5), then make a call to 
any sip phone or PSTN number.
2. SIP over WS  or SIP over WSS, has the same issue
3. Use sipml5 live demo page (http://sipml5.org/call.htm#) or test-call 
page(attached here) 

What is the expected output? What do you see instead?
After call setup successful, we expect two-way audio. But no audio at all.

(Instead of Asterisk, when tried with a webrtc2sip gateway, the chrome - sipml5 
client could make a call with audio).

What version of the product are you using? On what operating system?
sipml5 :Version 1.3.203
chrome : Version 30.0.1599.101
Asterisk: 11.5
OS : MAC OSX Vrsion 10.8.2

Please provide any additional information below.

Hi,
I try to use sipml5 with Asterisk(11.5) on chrome (30). I get the call 
established ( for ws and wss) between chrome and a sip phone (connecetd to 
asterisk), but no audio on both ends. My wireshark capture shows 
audio-udp(srtp) packets flowing from the asterisk to the chrome (no srtp 
packets from chrome). I use the PCMU or PCMA codec.

For the call initiated from Chrome, I get the following error:

SetRemoteDescription failed: Failed to update session state: ERROR_CONTENT 
SIPml-api.js:1
tsk_utils_log_error SIPml-api.js:1
tmedia_session_jsep01.onSetRemoteDescriptionError SIPml-api.js:3
(anonymous function)

For the incoming call to chrome, no errors.

Below is the log for outgoing call from chrome:

SIPML5 API version = 1.3.203 SIPml-api.js:1
User-Agent=Mozilla/5.0 (Macintosh; Intel Mac OS X 10_8_2) AppleWebKit/537.36 
(KHTML, like Gecko) Chrome/30.0.1599.101 Safari/537.36 SIPml-api.js:1
WebSocket supported = yes SIPml-api.js:1
Navigator friendly name = chrome SIPml-api.js:1
OS friendly name = mac SIPml-api.js:1
Have WebRTC = yes SIPml-api.js:1
Have GUM = yes SIPml-api.js:1
Engine initialized SIPml-api.js:1
s_websocket_server_url=wss://203.143.170.213:8089/ws SIPml-api.js:1
s_sip_outboundproxy_url=(null) SIPml-api.js:1
b_rtcweb_breaker_enabled=no SIPml-api.js:1
b_click2call_enabled=no SIPml-api.js:1
b_early_ims=yes SIPml-api.js:1
b_enable_media_stream_cache=no SIPml-api.js:1
o_bandwidth={} SIPml-api.js:1
o_video_size={} SIPml-api.js:1
SIP stack start: proxy='ns313841.ovh.net:11060', realm='<sip:asterisk>', 
impi='nicsec01110', impu='<sip:nicsec01110@203.143.170.213>' SIPml-api.js:1
Connecting to 'wss://203.143.170.213:8089/ws' SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
__tsip_transport_ws_onopen SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister 
SIPml-api.js:1
SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKmmSKI5PhXwDtetdCEaR59ZfKRZtebHDe;rport
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>
Contact: 
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws
s>;expires=200;click2call=no
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2347 REGISTER
Content-Length: 0
Max-Forwards: 70
Supported: path

 SIPml-api.js:1
session event = type = connecting - description = Connecting... 
wss_ast_call.html:42
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKmmSKI5Ph
XwDtetdCEaR59ZfKRZtebHDe
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>;tag=as77e33ffc
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2347 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="asterisk",nonce="6a604af9",stale=FALSE,algorithm=MD5

 SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 
SIPml-api.js:1
SEND: REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bK9RKzVx32PaGU0tfTe8h4Mf2NfJTeIWIO;rport
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>
Contact: 
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws
s>;expires=200;click2call=no
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2348 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="nicsec01110",realm="asterisk",nonce="6a604af9",uri="sip:asterisk",resp
onse="31dd1166570b0aeac2c8363299e2fe10",algorithm=MD5
Supported: path

 SIPml-api.js:1
session event = type = sent_request - description = REGISTER request 
successfully sent wss_ast_call.html:42
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bK9RKzVx32
PaGU0tfTe8h4Mf2NfJTeIWIO
From: <sip:nicsec01110@203.143.170.213>;tag=Rzglm2unvFJnGfqGzAw2
To: <sip:nicsec01110@203.143.170.213>;tag=as77e33ffc
Contact: 
<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=
200
Call-ID: d53c7b2c-0f5d-907c-a3f5-54d2907bea26
CSeq: 2348 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 24 Oct 2013 07:24:26 GMT;24

 SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js:1
session event = type = sent_request - description = REGISTER request 
successfully sent wss_ast_call.html:42
session event = type = connected - description = Connected wss_ast_call.html:42
State machine: c0000_Started_2_Outgoing_X_oINVITE SIPml-api.js:1
PeerConnectionClass = function RTCPeerConnection() { [native code] } 
SessionDescriptionClass = function RTCSessionDescription() { [native code] } 
IceCandidateClass = function RTCIceCandidate() { [native code] } SIPml-api.js:1
ICE 
servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.ne
t:3478"},{"url":"stun:numb.viagenie.ca:3478"}] SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
call event = connecting wss_ast_call.html:182
onGetUserMediaSuccess SIPml-api.js:1
createOffer SIPml-api.js:1
onCreateSdpSuccess SIPml-api.js:1
Uncaught TypeError: Cannot set property 'disabled' of null wss_ast_call.html:28
call event = m_stream_audio_local_added wss_ast_call.html:182
onSetLocalDescriptionSuccess SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
onIceCandidate = undefined SIPml-api.js:1
ICE GATHERING COMPLETED! SIPml-api.js:1
onIceGatheringCompleted SIPml-api.js:1
SEND: INVITE sip:20@203.143.170.213 SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKUk5hRO2K7XVEvbf4lGq1tGlVFLMLxSNC;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>
Contact: 
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=n
o;transport=wss>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15234 INVITE
Content-Type: application/sdp
Content-Length: 1525
Max-Forwards: 70

v=0
o=- 4518691561856059400 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
m=audio 62449 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 203.143.170.138
a=rtcp:62449 IN IP4 203.143.170.138
a=candidate:1315546826 1 udp 2113937151 203.143.170.138 62449 typ host 
generation 0
a=candidate:1315546826 2 udp 2113937151 203.143.170.138 62449 typ host 
generation 0
a=candidate:15358522 1 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=candidate:15358522 2 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=ice-ufrag:/MiMNlcyslcybRWu
a=ice-pwd:2keuIHyKr/pI9WPhrhL+Tc+s
a=ice-options:google-ice
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:lf04HXQ0nEA2Q1031haVgU50ZIZPIEamcbchSdZV
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:yUj0De1AXYe9kRXjCTd5ZsC7pcmVk2g/z8tQvoXr
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:107 CN/48000
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:3539336787 cname:Kdv90oeX0650TQ6k
a=ssrc:3539336787 msid:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV 
PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
a=ssrc:3539336787 mslabel:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
a=ssrc:3539336787 label:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
 SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKUk5hRO2K
7XVEvbf4lGq1tGlVFLMLxSNC
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as6423d2ee
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15234 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="asterisk",nonce="051134d3",stale=FALSE,algorithm=MD5

 SIPml-api.js:1
SEND: ACK sip:20@203.143.170.213 SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKUk5hRO2K7XVEvbf4lGq1tGlVFLMLxSNC;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as6423d2ee
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15234 ACK
Content-Length: 0
Max-Forwards: 70

 SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i401_407_INVITE SIPml-api.js:1
SEND: INVITE sip:20@203.143.170.213 SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKATMMli90AddZ5IIkBJihxvJBEgBB0iH2;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>
Contact: 
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=n
o;transport=wss>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Type: application/sdp
Content-Length: 1525
Max-Forwards: 70
Authorization: Digest 
username="nicsec01110",realm="asterisk",nonce="051134d3",uri="sip:20@203.143.170
.213",response="b16ffbfdf7321a04273e6bbe2ba5ba28",algorithm=MD5

v=0
o=- 4518691561856059400 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
m=audio 62449 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 203.143.170.138
a=rtcp:62449 IN IP4 203.143.170.138
a=candidate:1315546826 1 udp 2113937151 203.143.170.138 62449 typ host 
generation 0
a=candidate:1315546826 2 udp 2113937151 203.143.170.138 62449 typ host 
generation 0
a=candidate:15358522 1 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=candidate:15358522 2 tcp 1509957375 203.143.170.138 0 typ host generation 0
a=ice-ufrag:/MiMNlcyslcybRWu
a=ice-pwd:2keuIHyKr/pI9WPhrhL+Tc+s
a=ice-options:google-ice
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:lf04HXQ0nEA2Q1031haVgU50ZIZPIEamcbchSdZV
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:yUj0De1AXYe9kRXjCTd5ZsC7pcmVk2g/z8tQvoXr
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:107 CN/48000
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:3539336787 cname:Kdv90oeX0650TQ6k
a=ssrc:3539336787 msid:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV 
PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
a=ssrc:3539336787 mslabel:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DV
a=ssrc:3539336787 label:PNw4VZLOFS02RcFWopqWTO3zYGb8pGVgW3DVa0
 SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

 SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

 SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
call event = i_ao_request wss_ast_call.html:182
call event = i_ao_request wss_ast_call.html:182
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Length: 0
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

 SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
call event = i_ao_request wss_ast_call.html:182
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=51520;received=203.143.170.138;branch=z9hG4bKATMMli90
AddZ5IIkBJihxvJBEgBB0iH2
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact: <sip:20@203.143.170.213:5060;transport=WS>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 INVITE
Content-Type: application/sdp
Content-Length: 742
Server: Asterisk PBX 11.5.0
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

v=0
o=root 1449102145 1449102145 IN IP4 203.143.170.213
s=Asterisk PBX 11.5.0
c=IN IP4 203.143.170.213
t=0 0
m=audio 10016 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=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:662cde007895691646e75b180d979e91
a=ice-pwd:1446e3d55206319258de8afc0daae4df
a=candidate:Hcb8faad5 1 UDP 2130706431 203.143.170.213 10016 typ host
a=candidate:Ha0a046e 1 UDP 2130706431 10.10.4.110 10016 typ host
a=candidate:Hcb8faad5 2 UDP 2130706430 203.143.170.213 10017 typ host
a=candidate:Ha0a046e 2 UDP 2130706430 10.10.4.110 10017 typ host
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_80 
inline:vsKbW0mHuIZEMHKavHaeltWcT/75oSSkYJEVrNaT
 SIPml-api.js:1
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE SIPml-api.js:1
setRemoteDescription(answer)
v=0
o=root 1449102145 1449102145 IN IP4 203.143.170.213
s=Asterisk PBX 11.5.0
c=IN IP4 203.143.170.213
t=0 0
m=audio 10016 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=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:662cde007895691646e75b180d979e91
a=ice-pwd:1446e3d55206319258de8afc0daae4df
a=candidate:Hcb8faad5 1 UDP 2130706431 203.143.170.213 10016 typ host
a=candidate:Ha0a046e 1 UDP 2130706431 10.10.4.110 10016 typ host
a=candidate:Hcb8faad5 2 UDP 2130706430 203.143.170.213 10017 typ host
a=candidate:Ha0a046e 2 UDP 2130706430 10.10.4.110 10017 typ host
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_80 
inline:vsKbW0mHuIZEMHKavHaeltWcT/75oSSkYJEVrNaT
 SIPml-api.js:1
SEND: ACK sip:20@203.143.170.213:5060;transport=WS SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK7ain7moi7JrXk6Tqkmvq;rport
From: <sip:nicsec01110@203.143.170.213>;tag=WBqqgF4vaRrAHxxHVQq7
To: <sip:20@203.143.170.213>;tag=as7b9cc2ea
Contact: 
"undefined"<sips:nicsec01110@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=n
o;transport=wss>
Call-ID: ad2e715e-c6f8-f13c-ccc3-7a6adec876e2
CSeq: 15235 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="nicsec01110",realm="asterisk",nonce="051134d3",uri="sip:20@203.143.170
.213:5060;transport=WS",response="c7e81073d068c223d16b77eeae08eed4",algorithm=MD
5

 SIPml-api.js:1
onSetRemoteDescriptionError SIPml-api.js:1
SetRemoteDescription failed: Failed to update session state: ERROR_CONTENT 
SIPml-api.js:1
tsk_utils_log_error SIPml-api.js:1
tmedia_session_jsep01.onSetRemoteDescriptionError SIPml-api.js:3
(anonymous function) SIPml-api.js:3
call event = m_early_media wss_ast_call.html:182
call event = connected 

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Original issue reported on code.google.com by ath...@gmail.com on 25 Oct 2013 at 5:51

Attachments:

GoogleCodeExporter commented 8 years ago
Same here, on asterisk 11.6. Exactly same picture, onew-way RTP, no RTP from 
Chrome.

Original comment by zhega...@enaza.ru on 8 Nov 2013 at 10:37

GoogleCodeExporter commented 8 years ago
I have tested your script in this scenario, just replacing account and server 
info:

Chrome<--wss-->webrtc2sip<-->OpenSIPs<-->Asterisk<-->Voip_Provider<-->PSTN

and it doesn't work. Conclusion: Your script is totally broken! 

1)callSession = stack.newSession(,,,) is wrong. It has some missing elements 
like "sip_caps". 
2) what is audio-remote?
3) <audio id="audio_remote" autoplay="autoplay"> Is missing. 
4) Add it and rename  'audio-remote' by 'audio_remote in 2'
5) Your 'stack = new SIPml.Stack (,,) looks weird too. Don't assume anything, 
just provide all the parameters according to examples and documentation.

Because of that, this event 'm_stream_audio_local_added' never takes place.

After fixed, your script worked for me. Registration, Invite, bi-directional 
audio and BYE.

Cheers!

Miguel Oyarzo
Melbourne

Original comment by miguel.o...@medulla.com.au on 8 Nov 2013 at 2:22

GoogleCodeExporter commented 8 years ago
Miguel Oyarzo,
Thank you very much.
(I was away for a week ...)

What was missing  is "<audio id="audio_remote" autoplay="autoplay">" line
in the body .. and the audio_remote in the script!!!
AFTER THIS IT WORKS!
Thanks!!
(I just took the example script form the sipml5 test and tried to test it
.. I didn't look into the html lines .....)

(I removed some parameters  like sip_caps , for testing purposes ..  anyway
they are not making any impact on simple calls, at-least on my case ...)

I was testing with  sipml5 test page as well (http://sipml5.org/call.htm#)
but no audio ..

Finally audio worked, after patching the  Asterisk 11.5 code with the
latest patch ASTERISK-21383
(https://issues.asterisk.org/jira/browse/ASTERISK-21383)

Can you tell me, if Asterisk 11.5 alone worked for you or with the patch
ASTERISK-21383?

(And also, I've made the  wss call working with asterisk, as well ( without
webrtc2sip), after modifying asterisk code ....)

Thanks &

Rgds
Thava

Original comment by ath...@gmail.com on 13 Nov 2013 at 1:05

GoogleCodeExporter commented 8 years ago
This is my new scenario:
1) SipMl5 -> Asteriskx.x
2) SipMl5 -> Oversip--> Asteriskx.x

I have tested Asterisk 11.5, 11.6 and 11.7. In both scenarios I  got those 
errors:

WARNING[17942]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP 
policies
WARNING[17942]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP 
policies
WARNING[17942]: SRTP audio stream rejected, 'Could not set SRTP policies'

(res_srtp.so is loaded)

I'm passing this parameter to ./configure: --with-crypto --with-ssl=ssl 
--with-srtp

Is something missing?

Thanks,
Miguel
Melbourne

Original comment by miguel.o...@medulla.com.au on 28 Nov 2013 at 4:27

GoogleCodeExporter commented 8 years ago
Hello, I apply patch 21383 on asterisk 11.6 but no changes, no sound but call 
is established.
Use:
http://sipml5.org/call.htm#)
Disable Video: yes
WS ws://XX.XX.XX.XX:8088/ws
ICE servers [{ url: 'stun:stun.l.google.com:19302'}]
Disable 3GPP Early IMS:yes
Cache the media stream:yes
Disable Call button options:yes

Chrome Vesrison:31.0.1650.57 m

In the asterisk log 
 Stopping retransmission on
 of Request 125: Match Found

In Chrome Console I saw:
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport=55704;received=XX.XX.XX.XX;branch=z9hG4bKmNBSsCX42HK6
ZIWgugFApKDGT1HBgNPy

SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist

Original comment by Sitnik...@gmail.com on 28 Nov 2013 at 9:17

GoogleCodeExporter commented 8 years ago
This error usually means your SipML5 client is receiving SIP messages from a 
differnt path than ws/wss. I would think in SIP/routing issue from the end 
point.

Original comment by miguel.o...@medulla.com.au on 30 Nov 2013 at 12:19

GoogleCodeExporter commented 8 years ago
Same issue , not able hear voice from both endpoints
SIP user as follow :

[8001]
type=friend
host=dynamic
secret=XXXXXXXXXXXXXXX
context=default
icesupport = yes
insecure=port,invite
allow = h264,h263p,mpeg4

Original comment by ni...@unitedinfotech.in on 4 Jan 2014 at 12:28

GoogleCodeExporter commented 8 years ago
I've fixed the issue on my side.  If you give more details on the setup,
sip messages, colsole-outs.. I may be able to help ..

Original comment by ath...@gmail.com on 6 Jan 2014 at 3:43

GoogleCodeExporter commented 8 years ago
What asterisk version have you tested it on? 

We are using 11.7 with patches from:
https://issues.asterisk.org/jira/browse/ASTERISK-22961
https://issues.asterisk.org/jira/browse/ASTERISK-21930
https://issues.asterisk.org/jira/browse/ASTERISK-21383

And have no sound with packets flowing one way.

Original comment by Jidel...@gmail.com on 28 Jan 2014 at 10:21

GoogleCodeExporter commented 8 years ago
The issue has been fixed.

Original comment by Jidel...@gmail.com on 28 Jan 2014 at 11:59

GoogleCodeExporter commented 8 years ago
How you fixed this?

Original comment by hks1...@gmail.com on 6 Feb 2014 at 2:55

GoogleCodeExporter commented 8 years ago
Sorry, it wasn't fixed per se, but it works whenever there is no NAT. The 
question that remains, however, is how to make sure the signaling works (in a 
NAT situation). This seems to be a separate issue, which we are now trying to 
resolve. I will update the status of this ticket once we find a solution to 
that.

Original comment by Jidel...@gmail.com on 6 Feb 2014 at 3:01

GoogleCodeExporter commented 8 years ago
Why NAT is an issue for you? It is supposed that STUN provides visibility 
behind NAT. SIPml has default hard-code STUN servers, if you don't provide any 
in your javascript file (*stun.google* for example).  At the other end in 
asterisk you have to enable stun server too. 

Original comment by miguel.o...@medulla.com.au on 6 Feb 2014 at 3:15

GoogleCodeExporter commented 8 years ago
That's what we're trying to figure out. The STUN is added (google public) and 
that doesn't help. 

My rtp.conf contains:
icesupport=true
stunaddr=stun.l.google.com:19302

P.S. I'm not an expert so I may be mistaken in my conclusions. What I did 
notice though, is that one of my PCs (not behind nat) is able to make calls 
from both Firefox and Chrome over wss (with dtlsenable=yes), while for another 
pc it the calls are only possible over Chrome (with dtlsenable=no) or from 
Firefox (with dtlsenable=yes). So effectively that makes it a Chrome NAT 
problem...

Original comment by Jidel...@gmail.com on 6 Feb 2014 at 3:34

GoogleCodeExporter commented 8 years ago
I have here this error from Google Chrome 32 (Asterisk 11.6-cert:

tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. 

Original comment by sir.lo...@gmail.com on 17 Feb 2014 at 11:54

GoogleCodeExporter commented 8 years ago
Ok I've found the problem
If you compile Asterisk with OPUS support then there will be a conflict with 
ICE for some reason.
I should checkout the code of the OPUS Patch because it fails in Asterisk >11.6 
(i have it in Asterisk 11.5 and is fine)

Original comment by sir.lo...@gmail.com on 18 Feb 2014 at 12:47

GoogleCodeExporter commented 8 years ago
hello everyone,
i am using asterisk 11.7.0, sipml5 (www.sipml5.org/call.htm)

i am facing audio problems in (asterisk + sipml5 + chrome Version 27)
when i make calls from chrome browser to a sip phone connected to asterisk, i 
can see rtp packets flowing from sip phone to chrome but no rtp packets from 
chrome....

here is the log when i called from chrome to a sip phone, rtp is flowing in 
only one direction

    -- SIP/701-00000005 answered SIP/704-00000004
       > 0xef3150 -- Probation passed - setting RTP source address to 203.199.110.73:21592
Got  RTP packet from    203.199.110.73:21592 (type 00, seq 017916, ts 6646527, 
len 000160)
Sent RTP packet to      203.199.110.89:39240 (via ICE) (type 00, seq 007119, ts 
6646520, len 4294967284)
Got  RTP packet from    203.199.110.73:21592 (type 00, seq 017917, ts 6646767, 
len 000160)
Sent RTP packet to      203.199.110.89:39240 (via ICE) (type 00, seq 007120, ts 
6646760, len 4294967284)
Got  RTP packet from    203.199.110.73:21592 (type 00, seq 017918, ts 6646927, 
len 000160)
Sent RTP packet to      203.199.110.89:39240 (via ICE) (type 00, seq 007121, ts 
6646920, len 4294967284)
Got  RTP packet from    203.199.110.73:21592 (type 00, seq 017919, ts 6647087, 
len 000160)
Sent RTP packet to      203.199.110.89:39240 (via ICE) (type 00, seq 007122, ts 
6647080, len 4294967284)
Got  RTP packet from    203.199.110.73:21592 (type 00, seq 017920, ts 6647247, 
len 000160)
Sent RTP packet to      203.199.110.89:39240 (via ICE) (type 00, seq 007123, ts 
6647240, len 4294967284)

can anyone please help me with the solution...
thanx in advance

Original comment by shad.a.s...@gmail.com on 25 Feb 2014 at 6:38

GoogleCodeExporter commented 8 years ago
[deleted comment]
GoogleCodeExporter commented 8 years ago
I'm having a similar issue: established call but no audio on both ends, but I'm 
using freeswitch intead of asteriks.

Original comment by vasco...@gmail.com on 29 Apr 2014 at 10:08

GoogleCodeExporter commented 8 years ago
I've fixed the issue on my side.  If you give more details on the setup,
sip messages, colsole-outs.. I may be able to help ..

Original comment by ath...@gmail.com on 30 Apr 2014 at 5:57

GoogleCodeExporter commented 8 years ago
Ok, I'm posting the console outputs

Original comment by vasco...@gmail.com on 30 Apr 2014 at 2:19

GoogleCodeExporter commented 8 years ago
Ok, I am attaching the console output ... 
Is probably something missing in my implementation!

To receive a call my code is like that:
e.newSession.accept();

To make a call:
var makeCall = function(number){
                callSession = sipStack.newSession('call-audiovideo', {
                    video_remote: document.getElementById('video-remote'),
                    events_listener: { events: '*', listener: eventsListener } // optional: '*' means all events
                });
                callSession.call(number);
            } 

And I have a tag with the midia resource at the html file ...

Original comment by vasco...@gmail.com on 30 Apr 2014 at 3:53

Attachments:

GoogleCodeExporter commented 8 years ago
HI,

I am facing similar kind of problem. I am not able to receive the remote audio.
can anybody help me out.

As per my understanding some how I am unable to redirect my remote audio to my 
speaker.

Can any body please check my js code

Original comment by yogi....@gmail.com on 14 May 2014 at 1:04

Attachments: