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WRONG_CALL_STATE & wss port dead & freeSWITCH Process exits #1513
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are you for sure using the most recent sofia-sip library? |
Dear Engineer
I confirm that I am using version 1.13,Because I'm compiling , tip me :sofia-sip-ua >= 1.13… configure: error: no usable sofia-sip; please install sofia-sip-ua devel package or equivalent.
Thank you for your help!
![1641355197(1)](https://user-images.githubusercontent.com/30822672/148158773-b8cb23fc-106f-46b0-bd0a-1d7320e175b5.jpg)
…------------------ 原始邮件 ------------------
发件人: "signalwire/freeswitch" ***@***.***>;
发送时间: 2022年1月5日(星期三) 中午11:33
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主题: Re: [signalwire/freeswitch] WRONG_CALL_STATE & wss port dead & freeSWITCH Process exits (Issue #1513)
are you for sure using the most recent sofia-sip library?
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On the rather rare occasions that I see WRONG_CALL_STATE in the FS logs, I note that it's always preceded by "detaching session", but there is never the expected "Re-attaching to session" immediately following it as I see with successful calls. I also see the same 10 second timeout before the call is abandoned in WRONG_CALL_STATE. FS does not crash for me, that call simply fails. Then things behave normally after that. Hope this helps. |
Thank you for your reply. I still don't know how to solve the two problems
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I saw a small similarity with your issue. I sometimes (rarely) have a problem with Polycom SIP phones initiating a call to FS. When the call fails it shows WRONG_CALL_STATE after a 10 second timeout, same as your issue. I have very little information to guide me, but at first I thought sofia might be the common element. Now the lead developer, anthm, suggested that my phone is not responding to an authentication challenge from FS and to increase debug logging levels to see if this reveals the problem. Now I'm not sure if this is related to your issue as I have never worked with JSSIP nor web sockets, but maybe this can help you? Here is a link to his comment on the FreeSWITCH Slack channel: |
Lots of webrtc issues will be fixed by #2283 so I'd suggest trying that patch |
I am still encountering this problem, while using WSS (JsSIP) WebRTC for calls. Currently, I'm using FreeSWITCH Version 1.10.10-release+git20230813T165739Z4cb05e7f4a~64bit (git 4cb05e7 2023-08-13 16:57:39Z 64bit). I saw someone mentioning #2283 fixing a related issue, referencing sofia pull #233 in v1.13.17. I'm unsure about the sofia version I'm currently using, and I couldn't find a command in the documentation to check the sofia version. |
(1)WRONG_CALL_STATE
invite --->
407 <-----
ack --->
invite --->
the second invite,FreeSWITCH no have log print,10 seconds later, WRONG_CALL_STATE appears, By capturing the packet file, it can be seen that FreeSWITCH has received the invite for the second time.
(2)WSS port feigned death problem
After the problem occurs, I can Telnet, but the WSS handshake fails,Then I connect to fs_cli and execute command: reload mod_sofia
relaod fails, and the process exits. here the error logs and core file
(3) The FreeSWITCH process exits
Expected behavior
I used FreeSWITCH to convert webRTC to UDP, and I developed a WebrTC-based client bulk call myself using JSSIP.
Package version or git hash
Server Information:
virtual machine:
Trace logs
Provide freeswitch logs w/ DEBUG and UUID logging enabled
backtrace from core file
If applicable, provide the full backtrace from the core file.
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