SIP 0.7.5 + freeswitch 1.6.10 not working call from webpage on latest firefox/chrome

[macrom...@gmail.com]

Dear All;
I just started test sip.js and sipml5 from Oct 1. After fix lots of freeswitch things. The sip.js is still not working .

When I use "invite", it doesn't send any INVITE out. It stoped at "acquired local media streams" then nothing happened. I have enabled sipTrace:true. And following is the log.
When use sipml, it goes a bit further, it can call out and peer ringing, but immediately freeswith hang up with "codec negotiate error".
How can I overcome this problem and start using sip.js to make call from webrtc? thanks.


COnsole LoG:
Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.1.121 SIP/2.0
Via: SIP/2.0/WS cdnnh8fti21e.invalid;branch=z9hG4bK1887180
Max-Forwards: 70
To: <sip:10...@192.168.1.121>
From: <sip:10...@192.168.1.121>;tag=4th34ai2g0
Call-ID: o677k19uredmeisn0eovml
CSeq: 81 REGISTER
Contact: <sip:jjtv...@cdnnh8fti21e.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:a202fe2e-d993-4bf7-98dd-1bf6f7804bed>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, outbound
User-Agent: SIP.js/0.7.5
Content-Length: 0


sip-0.7.5.js (第 2884 行)

Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS cdnnh8fti21e.invalid;branch=z9hG4bK1887180;received=192.168.1.121;rport=34704
From: <sip:10...@192.168.1.121>;tag=4th34ai2g0
To: <sip:10...@192.168.1.121>;tag=Qv67jDe50r8Hr
Call-ID: o677k19uredmeisn0eovml
CSeq: 81 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.6.10~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
WWW-Authenticate: Digest realm="192.168.1.121", nonce="3889d87e-482a-4598-b52c-ee3272579d46", algorithm=MD5, qop="auth"
Content-Length: 0


sip-0.7.5.js (第 2884 行)

Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.1.121 SIP/2.0
Via: SIP/2.0/WS cdnnh8fti21e.invalid;branch=z9hG4bK7962978
Max-Forwards: 70
To: <sip:10...@192.168.1.121>
From: <sip:10...@192.168.1.121>;tag=4th34ai2g0
Call-ID: o677k19uredmeisn0eovml
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username="1001", realm="192.168.1.121", nonce="3889d87e-482a-4598-b52c-ee3272579d46", uri="sip:192.168.1.121", response="4f0f0bc41d526be96346cf1cc04879a7", qop=auth, cnonce="s26nistej88n", nc=00000001
Contact: <sip:jjtv...@cdnnh8fti21e.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:a202fe2e-d993-4bf7-98dd-1bf6f7804bed>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, outbound
User-Agent: SIP.js/0.7.5
Content-Length: 0


sip-0.7.5.js (第 2884 行)

Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS cdnnh8fti21e.invalid;branch=z9hG4bK7962978;received=192.168.1.121;rport=34704
From: <sip:10...@192.168.1.121>;tag=4th34ai2g0
To: <sip:10...@192.168.1.121>;tag=r5Z0m8y8X1y4K
Call-ID: o677k19uredmeisn0eovml
CSeq: 82 REGISTER
Contact: <sip:jjtv...@cdnnh8fti21e.invalid;transport=ws>;expires=600
Date: Tue, 04 Oct 2016 15:47:23 GMT
User-Agent: FreeSWITCH-mod_sofia/1.6.10~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0


sip-0.7.5.js (第 2884 行)

Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | received WebSocket text message:

NOTIFY sip:jjtv...@cdnnh8fti21e.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.121:5066;rport;branch=z9hG4bKN6eUyZX880S8r
Route: <sip:jjtv...@192.168.1.121:34704>;transport=ws
Max-Forwards: 70
From: <sip:10...@192.168.1.121>;tag=SeSSp3FcUaNQF
To: <sip:10...@192.168.1.121>
Call-ID: aeda3d63-04ec-1235-e1b1-9cd21eff3245
CSeq: 97461957 NOTIFY
Contact: <sip:mod_...@192.168.1.121:5060>
User-Agent: FreeSWITCH-mod_sofia/1.6.10~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Event: message-summary
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Subscription-State: terminated;reason=noresource
Content-Type: application/simple-message-summary
Content-Length: 65

Messages-Waiting: no
Message-Account: sip:10...@192.168.1.121


sip-0.7.5.js (第 2884 行)

Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | sending WebSocket message:

SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WS 192.168.1.121:5066;rport;branch=z9hG4bKN6eUyZX880S8r
To: <sip:10...@192.168.1.121>;tag=e90ls6pgln
From: <sip:10...@192.168.1.121>;tag=SeSSp3FcUaNQF
Call-ID: aeda3d63-04ec-1235-e1b1-9cd21eff3245
CSeq: 97461957 NOTIFY
Supported: outbound
User-Agent: SIP.js/0.7.5
Content-Length: 0


sip-0.7.5.js (第 2884 行)
Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transaction.nist | Timer J expired for non-INVITE server transaction z9hG4bKN6eUyZX880S8r
sip-0.7.5.js (第 2884 行)
Tue Oct 04 2016 23:47:25 GMT+0800 (CST) | sip.invitecontext.mediahandler | acquiring local media
sip-0.7.5.js (第 2884 行)
Tue Oct 04 2016 23:47:29 GMT+0800 (CST) | sip.invitecontext.mediahandler | acquired local media streams

[James Criscuolo]

Hi macromarship,

  Do you have any VPNs or extra network interfaces? These would add numerous ice candidates and make ice collection seemingly hang, or break. Using chrome://webrtc-internals/ would further confirm that. We also have a ua configuration parameter to cut down this time, but if you have invalid ice candidates it will not help.

James

On Tuesday, October 4, 2016 at 11:50:26 AM UTC-4, macrom...@gmail.com wrote:

Dear All;
   I just started test sip.js and sipml5 from Oct 1. After fix lots of freeswitch things. The sip.js is still not working .

   When I use "invite", it doesn't send any INVITE out. It stoped at "acquired local media streams" then nothing happened. I have enabled sipTrace:true. And following is the log.
   When use sipml, it goes a bit further, it can call out and peer ringing, but immediately freeswith hang up with "codec negotiate error".
   How can I overcome this problem and start using sip.js to make call from webrtc? thanks.


COnsole LoG:
Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.1.121 SIP/2.0
Via: SIP/2.0/WS cdnnh8fti21e.invalid;branch=z9hG4bK1887180
Max-Forwards: 70
To: <sip:10...@192.168.1.121>
From: <sip:10...@192.168.1.121>;tag=4th34ai2g0
Call-ID: o677k19uredmeisn0eovml
CSeq: 81 REGISTER

Contact: <sip:jjtv9msi@cdnnh8fti21e.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:a202fe2e-d993-4bf7-98dd-1bf6f7804bed>";expires=600

Contact: <sip:jjtv9msi@cdnnh8fti21e.invalid;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:a202fe2e-d993-4bf7-98dd-1bf6f7804bed>";expires=600

Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, outbound
User-Agent: SIP.js/0.7.5
Content-Length: 0


sip-0.7.5.js (第 2884 行)

Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS cdnnh8fti21e.invalid;branch=z9hG4bK7962978;received=192.168.1.121;rport=34704
From: <sip:10...@192.168.1.121>;tag=4th34ai2g0
To: <sip:10...@192.168.1.121>;tag=r5Z0m8y8X1y4K
Call-ID: o677k19uredmeisn0eovml
CSeq: 82 REGISTER

Contact: <sip:jjtv9msi@cdnnh8fti21e.invalid;transport=ws>;expires=600

Date: Tue, 04 Oct 2016 15:47:23 GMT
User-Agent: FreeSWITCH-mod_sofia/1.6.10~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0


sip-0.7.5.js (第 2884 行)

Tue Oct 04 2016 23:47:23 GMT+0800 (CST) | sip.transport | received WebSocket text message:

NOTIFY sip:jjtv9msi@cdnnh8fti21e.invalid;transport=ws SIP/2.0