[asterisk-bugs] [JIRA] (ASTERISK-26858) Got SIP response 500 "JsSIP Internal Error"

[Dragomir Haralambiev (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26858?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235704#comment-235704 ] 

Dragomir Haralambiev edited comment on ASTERISK-26858 at 3/12/17 5:28 PM:
--------------------------------------------------------------------------

Thanks for your quick reply.

When I report to JsSip (https://groups.google.com/forum/#!topic/jssip/MdO0-Y7-6fY) I receive follow answer:

====
The re-INVITE send by Asterisk (SIP re-invite Session-Timers) is wrong 
because it lacks the m=video line that was previously negotiated. 

This is a clear bug in Asterisk because it violates SDP rules on 
renegotiation. You should report it to Asterisk.
===== 

JsSip recommends referring to Asterisk , Asterisk recommends asking JsSip and problem is not resolved.

Maybe it is good idea to migrate from Asterisk to Freeswitch or Opensips?



was (Author: goup):
Thanks for your quick reply.

When I report to JsSip (https://groups.google.com/forum/#!topic/jssip/MdO0-Y7-6fY) I receive follow answer:

====
The re-INVITE send by Asterisk (SIP re-invite Session-Timers) is wrong 
because it lacks the m=video line that was previously negotiated. 

This is a clear bug in Asterisk because it violates SDP rules on 
renegotiation. You should report it to Asterisk.
===== 

JsSip recommends referring to Asterisk , Asterisk recommends asking JsSip and problem is not resolved.

Maybe it is good idea to migrate from Asterisk to Freeswitch ot Opensips?


> Got SIP response 500 "JsSIP Internal Error"
> -------------------------------------------
>
>                 Key: ASTERISK-26858
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26858
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/WebSocket
>    Affects Versions: 14.3.0
>         Environment: CentOS 7.3.1611, 
>            Reporter: Dragomir Haralambiev
>            Severity: Critical
>         Attachments: http.conf, sip.conf, tryit.jssip.net.log
>
>
> Hello,
> I try to test tryit.jssip.net with Asterisk 14.3.0.
> 1060 (WebRTC) call 1061 (SIP) using chan_sip.
> Call is connected, after one min sound is stopped in asterisk log is appear:
> Got SIP response 500 "JsSIP Internal Error" back from 192.168.1.104:5060
> If using SIPML5 all forking fine.
> What can I do to solve this problem?
> Here full asterisk log:
> == DTLS ECDH initialized (automatic), faster PFS enabled
> == Using SIP RTP CoS mark 5
> -- Executing [1061 at webrtc:1] Dial("SIP/1060-00000019", "SIP/1061") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/1061
> -- SIP/1061-0000001a is ringing
> > 0x7fbf14009110 -- Probation passed - setting RTP source address to 192.168.1.152:55334
> -- SIP/1061-0000001a answered SIP/1060-00000019
> -- Channel SIP/1061-0000001a joined 'simple_bridge' basic-bridge 
> -- Channel SIP/1060-00000019 joined 'simple_bridge' basic-bridge 
> > 0x7fbf14009110 -- Probation passed - setting RTP source address to 192.168.1.152:55334
> > 0x7fbec403a9e0 -- Probation passed - setting RTP source address to 192.168.1.104:53248
> -- Got SIP response 500 "JsSIP Internal Error" back from 192.168.1.104:5060
> > 0x7fbec403a9e0 -- Probation passed - setting RTP source address to 192.168.1.104:53248
> [Mar 10 21:08:11] WARNING[4801]: netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)



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