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versatica / tryit-jssip

New tryit-jssip application
https://tryit.jssip.net
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Got SIP response 500 "JsSIP Internal Error" back #5

Closed goup2010 closed 7 years ago

goup2010 commented 7 years ago

Hello,

I try to test tryit.jssip.net with Asterisk 14.3.0. 1060 (WebRTC) call 1061 (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.

Here full asterisk log: == DTLS ECDH initialized (automatic), faster PFS enabled == Using SIP RTP CoS mark 5 -- Executing [1061@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)

ibc commented 7 years ago

Sorry, this is JsSIP project, not Asterisk.

goup2010 commented 7 years ago

JSSIP is one part of communication. Normaly to send here problems.

ibc commented 7 years ago

Chrome/Firefox is also there, right? Did you report the same issue into the Chrome bug tracker?

To be 100% clear: We don't know your Asterisk configuration, we don't even know the current exact status of WebRTC in Asterisk . We just know that JsSIP properly deals with the WebRTC API provided by the browser and properly implements the SIP protocol. We are not responsible of how the different WebRTC engines (Chrome, Firefox, Asterisk, etc) deal one with each other.

Normaly to send here problems.

Yes, unfortunately is normal to get here all the reports about WebRTC issues with Asterisk. It just shouldn't be.

goup2010 commented 7 years ago

I apologize for the trouble.

ibc commented 7 years ago

You may want to ask in the JsSIP public mailing list about your problem with Asterisk, but we cannot keep it open here in the JsSIP issue tracker.