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JsSIP, the JavaScript SIP library
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Problem with hold() in JsSIP 0.4.1 #259

Closed axonaro closed 9 years ago

axonaro commented 9 years ago

When I try put call on hold , I get this

Fri Oct 24 2014 11:45:05 GMT+0400 (SAMT) | jssip.rtcsession | 29cddaf16957a30a2ead79aa7eace9a4@10.10.0.2:5060as648c43b8 | emitting event hold jssip-0.4.1.min.js:8 undefined Fri Oct 24 2014 11:45:05 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 29cddaf16957a30a2ead79aa7eace9a4@10.10.0.2:5060as648c43b8 | unable to set local description jssip-0.4.1.min.js:8 Fri Oct 24 2014 11:45:05 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 29cddaf16957a30a2ead79aa7eace9a4@10.10.0.2:5060as648c43b8 | Failed to set local offer sdp: Failed to push down transport description: Local fingerprint provided but no identity available. jssip-0.4.1.min.js:8

What I do wrong ?

PS Also I try mute/unmute and they work normally.

ibc commented 9 years ago

May you please describe your environment? Is there an Asterisk in there?

axonaro commented 9 years ago

debian 7 (wheezy) - Linux debian 2.6.32-5-amd64 #1 SMP Mon Oct 3 03:59:20 UTC 2011 x86_64 GNU/Linux Asterisk 11.12.0~dfsg-1~bpo70+1

Google Chrome 34.0.1847.137

ibc commented 9 years ago

hold/unhold is properly tested between JsSIP clients (through a SIP proxy). It does work.

If you have an Asterisk in the middle then the problem is in Asterisk. I've not tested how Asterisk behaves when it receives a re-INVITE within a WebRTC call. Please check the SDPs, and compare the SDP of the initial INVITE/200 and those in the re-INVITE/200. Sure you will find something broken in the SDP generated by Asterisk.

ibc commented 9 years ago

BTW, Asterisk 11 is from the past. We don't support here Asterisk, but even less Asterisk 11.

axonaro commented 9 years ago

how Asterisk behaves when it receives a re-INVITE within a WebRTC call.

But JsSIP not send re-INVITE to Asterisk

axonaro commented 9 years ago

hold/unhold is properly tested between JsSIP clients (through a SIP proxy)

Which sip proxy ? What version ? Thanks.

ibc commented 9 years ago

Then check the SDP in the reINVITE from Asterisk. Again, 11 is too old to expect it to work.

Proxies: Kamailio por OverSIP last versions.

sickpig commented 9 years ago

On Fri, Oct 24, 2014 at 11:48 AM, Iñaki Baz Castillo < notifications@github.com> wrote:

BTW, Asterisk 11 is from the past. We don't support here Asterisk, but even less Asterisk 11.

Sorry for jumping in but Asterisk 11 is not coming from the past. It's actively maintained the last release (11.13.1) happened on October 20th.

I do understand your frustration when tons of people report asterisk related problem here, though.

ibc commented 9 years ago

@sickpig I'm sorry. Did not know that. Anyhow in a hold operation there are two partida involved, in this case JsSIP and Asterisk. Assuming that the problem is always in JsSIP and always reporting that here (and never in Asterisk forums) indeed frustrates me.

ibc commented 9 years ago

partida -> parties (Android keyboard)

jmillan commented 9 years ago

@axonaro, anyway providing the SIP traces along with the rest of JsSIP log would make things easier for us to give any help.

http://jssip.net/documentation/0.4.x/api/ua_configuration_parameters/#parameter_trace_sip

axonaro commented 9 years ago

Sorry, but I don't understand you.

Then check the SDP in the reINVITE from Asterisk.

I click on button HOLD in my browser and JsSIP must send Re-INVITE to Asterisk.

ibc commented 9 years ago

Please enable traces and show the entire call logs and traces.

axonaro commented 9 years ago 
event.returnValue is deprecated. Please use the standard event.preventDefault() instead. jquery-1.10.2.min.js:5
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | configuration parameters after validation: jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · via_host: "9iab3v2ag6on.invalid" jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · password: NOT SHOWN jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · register_expires: 300 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · register: true jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · registrar_server: sip:my.server.com jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · ws_server_max_reconnection: 3 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · ws_server_reconnection_timeout: 4 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · connection_recovery_min_interval: 2 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · connection_recovery_max_interval: 30 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · use_preloaded_route: false jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · no_answer_timeout: 60000 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · stun_servers: ["stun:my.server.com:3478"] jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · turn_servers: [] jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · trace_sip: true jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · hack_via_tcp: false jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · hack_via_ws: false jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · hack_ip_in_contact: false jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · uri: sip:1999@my.server.com jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · ws_servers: [{"ws_uri":"ws://my.server.com:8088/ws","sip_uri":"<sip:my.server.com:8088;transport=ws;lr>","weight":0,"status":0,"scheme":"WS"}] jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · display_name: "1999" jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · instance_id: "994da86c-cd57-4bf9-a17a-5edddf51534b" jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · jssip_id: "v1irl" jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · hostport_params: "my.server.com" jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | · authorization_user: "1999" jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | user requested startup... jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | connecting to WebSocket ws://my.server.com:8088/ws jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event connecting jssip-0.4.1.min.js:8
Connecting to a non-secure WebSocket server from a secure origin is deprecated. jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | WebSocket ws://my.server.com:8088/ws connected jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | connection state set to 0 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event connected jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event newTransaction jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nict | z9hG4bK9264344 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | sending WebSocket message:

REGISTER sip:my.server.com SIP/2.0
Via: SIP/2.0/WS 9iab3v2ag6on.invalid;branch=z9hG4bK9264344
Max-Forwards: 69
To: <sip:1999@my.server.com>
From: "1999" <sip:1999@my.server.com>;tag=s1vi6gn7u0
Call-ID: trba63u4sfft907cc7ovbp
CSeq: 1 REGISTER
Contact: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:994da86c-cd57-4bf9-a17a-5edddf51534b>";expires=300
Expires: 300
Allow: ACK,CANCEL,BYE,OPTIONS,UPDATE,INVITE,MESSAGE
Supported: path,gruu,outbound
User-Agent: JsSIP 0.4.1
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 9iab3v2ag6on.invalid;branch=z9hG4bK9264344;received=192.168.200.157
From: "1999" <sip:1999@my.server.com>;tag=s1vi6gn7u0
To: <sip:1999@my.server.com>;tag=as7c4571f2
Call-ID: trba63u4sfft907cc7ovbp
CSeq: 1 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="my.server.com", nonce="6e7eaee7"
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nict | z9hG4bK9264344 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event newTransaction jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nict | z9hG4bK3134224 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | sending WebSocket message:

REGISTER sip:my.server.com SIP/2.0
Via: SIP/2.0/WS 9iab3v2ag6on.invalid;branch=z9hG4bK3134224
Max-Forwards: 69
To: <sip:1999@my.server.com>
From: "1999" <sip:1999@my.server.com>;tag=s1vi6gn7u0
Call-ID: trba63u4sfft907cc7ovbp
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="1999", realm="my.server.com", nonce="6e7eaee7", uri="sip:my.server.com", response="cbee3a207c1c47f3a70ba0d93fa2d798"
Contact: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:994da86c-cd57-4bf9-a17a-5edddf51534b>";expires=300
Expires: 300
Allow: ACK,CANCEL,BYE,OPTIONS,UPDATE,INVITE,MESSAGE
Supported: path,gruu,outbound
User-Agent: JsSIP 0.4.1
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nict | z9hG4bK9264344 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event transactionDestroyed jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | received WebSocket text message:

OPTIONS sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.10.0.2:5060;branch=z9hG4bK76e82d95
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.10.0.2>;tag=as1d712e48
To: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>
Contact: <sip:asterisk@10.10.0.2:5060;transport=WS>
Call-ID: 37cfd1db0566f5140b4905cd1612feb4@10.10.0.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 24 Oct 2014 10:50:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event newTransaction jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nist | z9hG4bK76e82d95 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 10.10.0.2:5060;branch=z9hG4bK76e82d95
To: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>;tag=7jsmlm81v0
From: "asterisk" <sip:asterisk@10.10.0.2>;tag=as1d712e48
Call-ID: 37cfd1db0566f5140b4905cd1612feb4@10.10.0.2:5060
CSeq: 102 OPTIONS
Allow: ACK,CANCEL,BYE,OPTIONS,UPDATE,INVITE,MESSAGE
Accept: application/sdp,application/dtmf-relay
Supported: outbound
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nist | z9hG4bK76e82d95 | Timer J expired for non-INVITE server transaction z9hG4bK76e82d95 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nist | z9hG4bK76e82d95 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event transactionDestroyed jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transport | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 9iab3v2ag6on.invalid;branch=z9hG4bK3134224;received=192.168.200.157
From: "1999" <sip:1999@my.server.com>;tag=s1vi6gn7u0
To: <sip:1999@my.server.com>;tag=as7c4571f2
Call-ID: trba63u4sfft907cc7ovbp
CSeq: 2 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>;expires=300
Date: Fri, 24 Oct 2014 10:50:19 GMT
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nict | z9hG4bK3134224 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event registered jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.transaction.nict | z9hG4bK3134224 | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:19 GMT+0400 (SAMT) | jssip.ua | emitting event transactionDestroyed jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.transport | received WebSocket text message:

INVITE sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.10.0.2:5060;branch=z9hG4bK6ddd113a;rport
Max-Forwards: 70
From: "Гордеев Александр" <sip:1424@10.10.0.2>;tag=as19937212
To: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>
Contact: <sip:1424@10.10.0.2:5060;transport=WS>
Call-ID: 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 24 Oct 2014 10:50:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1132

v=0
o=Cisco-SIPUA 1891248673 1891248673 IN IP4 10.10.0.2
s=SIP Call
c=IN IP4 10.10.0.2
t=0 0
m=audio 16228 RTP/SAVPF 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:368666eb4a6c723a5046c5f35b859c1f
a=ice-pwd:64bcf36a07285a256cac1c0f57de54ff
a=candidate:Hd5f83f86 1 UDP 2130706431 213.48.163.34 16228 typ host
a=candidate:Hd5dbf48c 1 UDP 2130706431 213.19.44.40 16228 typ host
a=candidate:Hd5dbf48f 1 UDP 2130706431 213.19.44.43 16228 typ host
a=candidate:Ha0a0002 1 UDP 2130706431 10.10.0.2 16228 typ host
a=candidate:Hac100002 1 UDP 2130706431 172.16.0.2 16228 typ host
a=candidate:Hd5f83f86 2 UDP 2130706430 213.48.163.34 16229 typ host
a=candidate:Hd5dbf48c 2 UDP 2130706430 213.19.44.40 16229 typ host
a=candidate:Hd5dbf48f 2 UDP 2130706430 213.19.44.43 16229 typ host
a=candidate:Ha0a0002 2 UDP 2130706430 10.10.0.2 16229 typ host
a=candidate:Hac100002 2 UDP 2130706430 172.16.0.2 16229 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Me35NFAt1DKZgGBq6/8BiXc0u02ua3GNNIpcxsS+

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.ua | emitting event newTransaction jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.transport | sending WebSocket message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS 10.10.0.2:5060;branch=z9hG4bK6ddd113a;rport
To: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>
From: "Гордеев Александр" <sip:1424@10.10.0.2>;tag=as19937212
Call-ID: 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060
CSeq: 102 INVITE
Supported: outbound
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.dialog | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060fc6va4po32as19937212 | new UAS dialog created with status EARLY jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | stream added: default jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.ua | emitting event newRTCSession jssip-0.4.1.min.js:8
Uncaught TypeError: Cannot read property 'document' of null jquery-1.10.2.min.js:5
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.transport | sending WebSocket message:

SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.10.0.2:5060;branch=z9hG4bK6ddd113a;rport
To: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>;tag=fc6va4po32
From: "Гордеев Александр" <sip:1424@10.10.0.2>;tag=as19937212
Call-ID: 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060
CSeq: 102 INVITE
Contact: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>
Supported: outbound
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:24 GMT+0400 (SAMT) | jssip.rtcsession | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | emitting event progress jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.dialog | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060fc6va4po32as19937212 | dialog 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060fc6va4po32as19937212  changed to CONFIRMED state jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | requesting access to local media jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | got local media stream jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | emitting event connecting jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | ICE connection state changed to "checking" jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | ICE candidate received: a=candidate:579488925 1 udp 2122260223 192.168.200.157 53831 typ host generation 0
 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | ICE candidate received: a=candidate:579488925 2 udp 2122260222 192.168.200.157 37890 typ host generation 0
 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | ICE candidate received: a=candidate:1812574317 1 tcp 1518280447 192.168.200.157 0 typ host generation 0
 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | ICE candidate received: a=candidate:1812574317 2 tcp 1518280446 192.168.200.157 0 typ host generation 0
 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.transaction.ist | z9hG4bK6ddd113a | emitting event stateChanged jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.transport | sending WebSocket message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 10.10.0.2:5060;branch=z9hG4bK6ddd113a;rport
To: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>;tag=fc6va4po32
From: "Гордеев Александр" <sip:1424@10.10.0.2>;tag=as19937212
Call-ID: 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060
CSeq: 102 INVITE
Contact: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>
X-Foo: foo
X-Bar: bar
Supported: outbound
Content-Type: application/sdp
Content-Length: 1042

v=0
o=- 3186128285565709178 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS RpPIVOxeOy2E0iho3XI3HKpbZvXPn9StsYt1
m=audio 53831 RTP/SAVPF 8 101
c=IN IP4 192.168.200.157
a=rtcp:37890 IN IP4 192.168.200.157
a=candidate:579488925 1 udp 2122260223 192.168.200.157 53831 typ host generation 0
a=candidate:579488925 2 udp 2122260222 192.168.200.157 37890 typ host generation 0
a=candidate:1812574317 1 tcp 1518280447 192.168.200.157 0 typ host generation 0
a=candidate:1812574317 2 tcp 1518280446 192.168.200.157 0 typ host generation 0
a=ice-ufrag:O7BqZZVjTF1inA3i
a=ice-pwd:YRUIPzsAP2RuM/JbhCuAA6cR
a=mid:audio
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:P2j+BYI5VrKVTzBAywqfhJX+veJAXPpKlCcfjWJG
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:3283143758 cname:kuWKxarBqwdrzcEf
a=ssrc:3283143758 msid:RpPIVOxeOy2E0iho3XI3HKpbZvXPn9StsYt1 5a37493b-3eef-4ba2-adc8-23b9b06552f0
a=ssrc:3283143758 mslabel:RpPIVOxeOy2E0iho3XI3HKpbZvXPn9StsYt1
a=ssrc:3283143758 label:5a37493b-3eef-4ba2-adc8-23b9b06552f0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | emitting event accepted jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.transport | received WebSocket text message:

ACK sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.10.0.2:5060;branch=z9hG4bK74418b1b;rport
Max-Forwards: 70
From: "Гордеев Александр" <sip:1424@10.10.0.2>;tag=as19937212
To: <sip:cf7a65c8@9iab3v2ag6on.invalid;transport=ws>;tag=fc6va4po32
Contact: <sip:1424@10.10.0.2:5060;transport=WS>
Call-ID: 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | emitting event confirmed jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:26 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | ICE connection state changed to "connected" jssip-0.4.1.min.js:8
JsSIPcurSession.isOnHold();
Object {local: false, remote: false}
JsSIPcurSession.hold();
Fri Oct 24 2014 14:50:50 GMT+0400 (SAMT) | jssip.rtcsession | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | emitting event hold jssip-0.4.1.min.js:8
undefined
Fri Oct 24 2014 14:50:50 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | unable to set local description jssip-0.4.1.min.js:8
Fri Oct 24 2014 14:50:50 GMT+0400 (SAMT) | jssip.rtcsession.rtcmediahandler | 3323a7011f68faa7176fe7e4430c86c8@10.10.0.2:5060as19937212 | Failed to set local offer sdp: Failed to push down transport description: Local fingerprint provided but no identity available.
ibc commented 9 years ago

This is the INVITE received from your Asterisk:

v=0
o=Cisco-SIPUA 1891248673 1891248673 IN IP4 10.10.0.2
s=SIP Call
c=IN IP4 10.10.0.2
t=0 0
m=audio 16228 RTP/SAVPF 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:368666eb4a6c723a5046c5f35b859c1f
a=ice-pwd:64bcf36a07285a256cac1c0f57de54ff
a=candidate:Hd5f83f86 1 UDP 2130706431 213.48.163.34 16228 typ host
a=candidate:Hd5dbf48c 1 UDP 2130706431 213.19.44.40 16228 typ host
a=candidate:Hd5dbf48f 1 UDP 2130706431 213.19.44.43 16228 typ host
a=candidate:Ha0a0002 1 UDP 2130706431 10.10.0.2 16228 typ host
a=candidate:Hac100002 1 UDP 2130706431 172.16.0.2 16228 typ host
a=candidate:Hd5f83f86 2 UDP 2130706430 213.48.163.34 16229 typ host
a=candidate:Hd5dbf48c 2 UDP 2130706430 213.19.44.40 16229 typ host
a=candidate:Hd5dbf48f 2 UDP 2130706430 213.19.44.43 16229 typ host
a=candidate:Ha0a0002 2 UDP 2130706430 10.10.0.2 16229 typ host
a=candidate:Hac100002 2 UDP 2130706430 172.16.0.2 16229 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Me35NFAt1DKZgGBq6/8BiXc0u02ua3GNNIpcxsS+

As you see there is no a=fingerprint line which means that it is NOT using DTLS but SDES. Chrome somehow still allows it but it is deprecated and dropped from the WebRTC specs, so it is an invalid SDP and thus, regardless the initial INVITE works, next stuff fails because Chrome does internally expect to find DTLS when doing a new SDP re-negotiation.

Sorry, this is an Asterisk configuration issue, fix that by enabling DTLS (we cannot help in how to do that).

axonaro commented 9 years ago

Thank you for your patience and help.

ibc commented 9 years ago

np, let's us now if it works after enabling DTLS in Asterisk. Not sure if this works, but search for "dtls" words on it: http://sipjs.com/guides/server-configuration/asterisk/

axonaro commented 9 years ago

Of course, if I decide this problem, then I describe the solution here

sickpig commented 9 years ago

if you're in a hurry just disable DTLS on the browser side using:

'DtlsSrtpKeyAgreement': 'false'

in the rtc media handler constraint.

mind you sooner or later chrome will drop SDES support, so it should be better to follow @ibc and setting up DTLS on the asterisk side.

On Fri, Oct 24, 2014 at 1:24 PM, axonaro notifications@github.com wrote:

Of course, if I decide this problem, then I describe the solution here

— Reply to this email directly or view it on GitHub https://github.com/versatica/JsSIP/issues/259#issuecomment-60374743.

ibc commented 9 years ago

@sickpig for what I've seen, that will not prevent the renegotiation issue in which, somehow, it does expect DTLS. Not sure, won't check it anyhow :)

sickpig commented 9 years ago

On Fri, Oct 24, 2014 at 1:53 PM, Iñaki Baz Castillo < notifications@github.com> wrote:

@sickpig https://github.com/sickpig for what I've seen, that will not prevent the renegotiation issue in which, somehow, it does expect DTLS. Not sure, won't check it anyhow :)

@ibc now that I think about it I think you're right, maybe if time it's not a constraint he should just give a quick try and see what will happen

— Reply to this email directly or view it on GitHub https://github.com/versatica/JsSIP/issues/259#issuecomment-60377142.

axonaro commented 9 years ago

if it works after enabling DTLS in Asterisk

It worked ! Thanks !!

ibc commented 9 years ago

Great :)