search

jarsy / sipml5

Automatically exported from code.google.com/p/sipml5
BSD 3-Clause "New" or "Revised" License
0 stars 0 forks source link

No Audio at all after updating to latest Asterisk Patch #38

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago 
a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?
1. Installed Asterisk from scratch from the Doubango page, and downloaded 
latest patch and installed, completed all steps to get Asterisk running.

2. Copied the sip.conf, extensions.conf, http.conf, and users.conf from the 
repository.

3. Added avpf=yes and encryption=yes to user 1061 to make call with Chrome.

4. Registered both users, one on Mac chrome and one on Ubuntu Chrome. (Same 
behavior on Windows XP as well)

5. I can call and connect, but I get no audio at all on either end, and I see 
the following logs in Asterisk. Note the error, which I've added whitespace 
around:

  == Using SIP RTP CoS mark 5
    -- Executing [101@default:1] Dial("SIP/1060-00000002", "SIP/1061") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1061
    -- SIP/1061-00000003 is ringing

[Sep 14 00:20:49] ERROR[7196][C-00000002]: netsock2.c:269 ast_sockaddr_resolve: 
getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known

[Sep 14 00:20:49] WARNING[7196][C-00000002]: chan_sip.c:15200 
__set_address_from_contact: Invalid host name in Contact: (can't resolve in 
DNS) : 'df7jal23ls0d.invalid'
    -- SIP/1061-00000003 answered SIP/1060-00000002
       > User Agent transport = WS�       > User Agent transport = WS
�  == Spawn extension (default, 101, 1) exited non-zero on 'SIP/1060-00000002'
       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS�       > User Agent transport = WS
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).

What is the expected output? What do you see instead?

The expected output is to hear audio. But there is no audio.

What version of the product are you using? On what operating system?

I'm on Asterisk r372699M. I checked out fresh at the advice of Mamadou. I 
downloaded the latest patch as of the time of this post and applied it. The 
Asterisk server is on Ubuntu 11.04 64 bit, Chrome 23.0.1266.0 is on Ubuntu 
12.04 64 bit, and I have a Chrome Canary 23.0.1265.0 on both Mac 10.7 and 
Windows XP. 

Please provide any additional information below.

I tested with 2 Chrome's because the call won't even ring to XLite on Mac 10.7. 
 Before the update, I had one way audio. from Chrome to XLite only.

Again, I'm using all of the defaults from the repository and followed the docs 
exactly. Please let me know if you need more info.

JavaScript from Chrome:

State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js:55
__on_state_change tsk_utils.js:55
Media Added tsk_utils.js:55
Call in progress... tsk_utils.js:55
SEND: INVITE sip:101@james.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKAWSBi1XkVwt90MvI0pZV29kR5xFCEfaz;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55418 INVITE
Content-Type: application/sdp
Content-Length: 1282
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKAWSBi1XkVwt90MvI0pZV
29kR5xFCEfaz
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as2b85ef53
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55418 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="doubango.org",nonce="4e408110",stale=FALSE,algorithm=MD5

 tsk_utils.js:55
SEND: ACK sip:101@james.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKAWSBi1XkVwt90MvI0pZV29kR5xFCEfaz;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as2b85ef53
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55418 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js:55
SEND: INVITE sip:101@james.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKgwSmiD1S3ffO84uAIxAatvAFk9cAdJFt;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Type: application/sdp
Content-Length: 1282
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:101@james.org",re
sponse="da920b16ba2d972a4b955f64d6be5b72",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKgwSmiD1S3ffO84uAIxAa
tvAFk9cAdJFt
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>
Contact: <sip:101@10.168.1.6:5060;transport=WS>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Trying tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKgwSmiD1S3ffO84uAIxAa
tvAFk9cAdJFt
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Contact: <sip:101@10.168.1.6:5060;transport=WS>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

 tsk_utils.js:55
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:55
Ringing tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKgwSmiD1S3ffO84uAIxAa
tvAFk9cAdJFt
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Contact: <sip:101@10.168.1.6:5060;transport=WS>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 INVITE
Content-Type: application/sdp
Content-Length: 650
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

v=0
o=root 426452035 426452035 IN IP4 10.168.1.6
s=Asterisk PBX SVN-trunk-r372699M
c=IN IP4 10.168.1.6
t=0 0
m=audio 10650 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:27b11bac1e57e2242d92cb8b2e65b422
a=ice-pwd:21b0dc964ac3faf11d5e9ee05b4feff0
a=candidate:Haa89741 1 udp 2130706431 10.168.1.6 10650 typ host generation 0 
svn 10
a=candidate:Haa89741 2 udp 2130706430 10.168.1.6 10651 typ host generation 0 
svn 10
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:gFgZbuFJl3XlTE8xviOwzMWnGGy4qaNMMQYZ45GR
 tsk_utils.js:55
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js:55
__on_open tsk_utils.js:55
__on_state_change tsk_utils.js:55
SEND: ACK sip:101@10.168.1.6:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB7KDOPzXz6iPArF1flC7;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55419 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:101@10.168.1.6:50
60;transport=WS",response="d4455e5ab99c3f7a961d67c5f6d464ea",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

 tsk_utils.js:55
OK tsk_utils.js:55
In Call tsk_utils.js:55
SEND: INVITE sip:101@10.168.1.6:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKJnaoITOlWLPF5lCKomEKz68BJemWAsCl;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55420 INVITE
Content-Type: application/sdp
Content-Length: 3104
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:101@10.168.1.6:50
60;transport=WS",response="6d618b54128182514ba7259239d7416a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 2 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
m=video 32828 RTP/SAVPF 100 101 102
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
a=rtpmap:100 VP8/90000
a=rtpmap:101 red/90000
a=rtpmap:102 ulpfec/90000
a=ssrc:3925540917 cname:LvYPKr+O3jVL3995
a=ssrc:3925540917 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:3925540917 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS10
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKJnaoITOlWLPF5lCKomEK
z68BJemWAsCl
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55420 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js:55
SEND: ACK sip:101@10.168.1.6:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKJnaoITOlWLPF5lCKomEKz68BJemWAsCl;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55420 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js:55
SEND: INVITE sip:101@10.168.1.6:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKy7uaxTPmuiFOT1S4hMFg3Nt5b6nLkxXp;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Contact: "1060"<sip:1060@df7jal23ls0d.invalid;transport=ws>;+sip.ice
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55421 INVITE
Content-Type: application/sdp
Content-Length: 3104
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:101@10.168.1.6:50
60;transport=WS",response="6d618b54128182514ba7259239d7416a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

v=0
o=- 126270660 3 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 32828 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
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=ssrc:2823364993 cname:LvYPKr+O3jVL3995
a=ssrc:2823364993 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:2823364993 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS00
m=video 32828 RTP/SAVPF 100 101 102
c=IN IP4 50.43.1.8
a=rtcp:32828 IN IP4 50.43.1.8
a=candidate:4086812097 1 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:4086812097 2 udp 2130714367 10.3.1.1 46941 typ host generation 0
a=candidate:2971284297 1 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:2971284297 2 udp 1912610559 50.43.1.8 32828 typ srflx generation 0
a=candidate:3172217137 1 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=candidate:3172217137 2 tcp 1694506751 10.3.1.1 60049 typ host generation 0
a=ice-ufrag:LLwIc5JWkKcHQGiA
a=ice-pwd:Ejj4ueILoSFLKIPhnU+blZ6R
a=sendrecv
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:C7nOKxVL0L/pK8ZYIbtH2mTE0GdlXUSmzdeT5iHp
a=rtpmap:100 VP8/90000
a=rtpmap:101 red/90000
a=rtpmap:102 ulpfec/90000
a=ssrc:3925540917 cname:LvYPKr+O3jVL3995
a=ssrc:3925540917 mslabel:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS
a=ssrc:3925540917 label:HHk5sD0ZayL5nQaeZsWHmOmTBn2gndFJ6NLS10
 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKy7uaxTPmuiFOT1S4hMFg
3Nt5b6nLkxXp
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55421 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js:55
SEND: ACK sip:101@10.168.1.6:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKy7uaxTPmuiFOT1S4hMFg3Nt5b6nLkxXp;rport
From: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
To: <sip:101@james.org>;tag=as7e61952f
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 55421 ACK
Content-Length: 0
Max-Forwards: 70

 tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=BYE sip:1060@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK31bf63fb
From: <sip:101@james.org>;tag=as7e61952f
To: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 102 BYE
Content-Length: 0
Max-Forwards: 70
User-Agent: Asterisk PBX SVN-trunk-r372699M
Proxy-Authorization: Digest 
username="1060",realm="doubango.org",nonce="4e408110",uri="sip:james.org",respon
se="9f3bb2fa621b20d320034f888e4580c6",algorithm=MD5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

 tsk_utils.js:55
State machine: x0000_Any_2_Terminated_X_iBYE tsk_utils.js:55
=== INVITE Dialog terminated === tsk_utils.js:55
__on_state_change tsk_utils.js:55
PeerConnection::stop() tsk_utils.js:55
SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK31bf63fb
From: <sip:101@james.org>;tag=as7e61952f
To: <sip:1060@james.org>;tag=ACODOuXTJzNKEwrv9ip8
Contact: <sip:1060@df7jal23ls0d.invalid;transport=ws>
Call-ID: 467f376f-8129-6993-4b96-3e4678df2eb9
CSeq: 102 BYE
Content-Length: 0

 tsk_utils.js:55
Call terminated tsk_utils.js:55
2Media Removed tsk_utils.js:55
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister 
tsk_utils.js:55
SEND: REGISTER sip:james.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKa5GXC5sZp7be9ufmtyK1DVCnbJtakeDV;rport
From: <sip:1060@james.org>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:1060@james.org>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;transport=ws>;expires=200;+g.oma.sip-im;+au
dio;language="en,fr"
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52213 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="55a3ba48",uri="sip:james.org",respon
se="45f027d273101c3f3d0e8e775d2020ce",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

 tsk_utils.js:55
REGISTER request successfully sent tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKa5GXC5sZp7be9ufmtyK1
DVCnbJtakeDV
From: <sip:1060@james.org>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:1060@james.org>;tag=as029056b1
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52213 REGISTER
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest 
realm="doubango.org",nonce="129f1225",stale=FALSE,algorithm=MD5

 tsk_utils.js:55
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 
tsk_utils.js:55
SEND: REGISTER sip:james.org SIP/2.0
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;branch=z9hG4bKtciI7O8LAEn5LUX97Nut4Z5SG3SkhuDy;rport
From: <sip:1060@james.org>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:1060@james.org>
Contact: 
"1060"<sip:1060@df7jal23ls0d.invalid;transport=ws>;expires=200;+g.oma.sip-im;+au
dio;language="en,fr"
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52214 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest 
username="1060",realm="doubango.org",nonce="129f1225",uri="sip:james.org",respon
se="13c2126a4e68e9282761a9b171edc1bc",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.0.89.0/
Organization: Doubango Telecom

 tsk_utils.js:55
REGISTER request successfully sent tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;received=50.43.1.8;branch=z9hG4bKtciI7O8LAEn5LUX97Nut
4Z5SG3SkhuDy
From: <sip:1060@james.org>;tag=9ZR1Q4e8UHtJg5X1NcfX
To: <sip:1060@james.org>;tag=as029056b1
Contact: <sip:1060@df7jal23ls0d.invalid;transport=ws>;expires=200
Call-ID: ff548533-c2de-a693-bcd2-b99766c615c5
CSeq: 52214 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r372699M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: Fri, 14 Sep 2012 0:40:53 GMT

 tsk_utils.js:55
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js:55
__tsip_transport_ws_onmessage tsk_utils.js:55
recv=NOTIFY sip:1060@df7jal23ls0d.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK7b067980
From: "asterisk"<sip:asterisk@10.168.1.6>;tag=as0660ad45
To: <sip:1060@df7jal23ls0d.invalid;transport=ws>
Contact: <sip:asterisk@10.168.1.6:5060;transport=WS>
Call-ID: 179abe547b60ece40040e4bd72eccd83@10.168.1.6:5060
CSeq: 102 NOTIFY
Content-Type: application/simple-message-summary
Content-Length: 106
Max-Forwards: 70
User-Agent: Asterisk PBX SVN-trunk-r372699M
Event: message-summary

Messages-Waiting: no
Message-Account: sip:asterisk@10.168.1.6;transport=WS
Voice-Message: 0/0 (0/0)
 tsk_utils.js:55
SEND: SIP/2.0 481 Dialog/Transaction Does Not Exist
Via: SIP/2.0/WS 10.168.1.6:5060;branch=z9hG4bK7b067980
From: "asterisk"<sip:asterisk@10.168.1.6>;tag=as0660ad45
To: <sip:1060@df7jal23ls0d.invalid;transport=ws>
Call-ID: 179abe547b60ece40040e4bd72eccd83@10.168.1.6:5060
CSeq: 102 NOTIFY
Content-Length:

Original issue reported on code.google.com by james.mo...@a-cti.com on 14 Sep 2012 at 12:51

GoogleCodeExporter commented 9 years ago 
I looks like your Asterisk server is running on private network (not public). 
Is it right?

Original comment by boss...@yahoo.fr on 14 Sep 2012 at 4:02

GoogleCodeExporter commented 9 years ago 
It's on an EC2 server, and it's publicly accessible. As for the Chrome clients, 
they are of course behind NAT.

Original comment by james.mo...@a-cti.com on 14 Sep 2012 at 4:55

GoogleCodeExporter commented 9 years ago 
[deleted comment]
GoogleCodeExporter commented 9 years ago 
Update on this issue:  I installed Asterisk 372699 and applied the patch on my 
laptop, which runs Ubuntu 12.04 64 bit.  I first tried with the configuration 
used in http://code.google.com/p/sipml5/wiki/Asterisk#Building_source_code and 
then experimented with some of the settings like nat, icecupport, tcpenabled, 
and other various settings.

I installed on my local PC to try and eliminate NAT as an issue. I was hoping 
that having the 2 Chrome clients and Asterisk behind the same private network 
might help.  It didn't. I get the same error as originally reported:

[Sep 14 00:20:49] ERROR[7196][C-00000002]: netsock2.c:269 ast_sockaddr_resolve: 
getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known

I also want to point out that this was a bug I reported to the Asterisk issue 
tracker. The bug was fixed in revision 371483. Is this bug back, or is 
something else going on? I'm considering reporting this to the Asterisk mailing 
list as well. What do you think?

https://issues.asterisk.org/jira/browse/ASTERISK-20238

Original comment by james.mo...@a-cti.com on 20 Sep 2012 at 9:58

GoogleCodeExporter commented 9 years ago 
hi every one .. 
is still no solution for this problem !!?

#############
getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): No address associated with  
       hostname
#############

Original comment by soufiane...@gmail.com on 27 Feb 2014 at 4:15

GoogleCodeExporter commented 9 years ago 
This issue seems no solution.

Original comment by joseph...@gmail.com on 17 May 2014 at 10:31