[asterisk-bugs] [JIRA] (ASTERISK-27587) Asterisk webrtc con JSSIP dont close connection

Kevin Harwell (JIRA) noreply at issues.asterisk.org
Mon Jan 15 15:54:49 CST 2018


    [ https://issues.asterisk.org/jira/browse/ASTERISK-27587?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=241599#comment-241599 ] 

Kevin Harwell commented on ASTERISK-27587:
------------------------------------------

Per the Asterisk versions page [1], the maintenance (bug fix) support for the Asterisk branch you are using has ended. For continued maintenance support please move to a supported branch of Asterisk. After testing with a supported branch, if you find this problem has not been resolved, please open a new issue against the latest version of that Asterisk branch.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions



> Asterisk webrtc con JSSIP dont close connection
> -----------------------------------------------
>
>                 Key: ASTERISK-27587
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27587
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Core/HTTP
>    Affects Versions: 14.5.0, 14.6.2
>            Reporter: VIP2PHONE INC
>              Labels: webrtc
>
> Greetings, we currently have the version of Asterisk 14.5.0 implemented, we use JSSIP as the base library for the Webrct webphone, initially it works correctly with a small number of extensions, but when we use a pbx with 100 extensions the response on port 8089 tcp of webrtc denies the connections, and the connections that existed begin to be lost.
> We observe in the system message:
> TCP: request_sock_TCP: Possible SYN flooding on port 8089. Sending cookies. Check SNMP counters.
> Log Asterisk
> http.c: HTTP session count exceeded 100 sessions.
> We make the adjustment in the http.conf file
> sessionlimit = 500
> But the message Greetings, we currently have the version of Asterisk 14.5.0 implemented, we use JSSIP as the base library for the Webrct webphone, initially it works correctly with a small amount of extensions, but when we use a pbx with 100 extensions the answer on port 8089 tcp of webrtc denies the connections, and the connections that existed begin to get lost.
> We observe in the system message:
> TCP: request_sock_TCP: Possible SYN flooding on port 8089. Sending cookies. Check SNMP counters.
> We modify
> echo "4096"> / proc / sys / net / ipv4 / tcp_max_syn_backlog
> Log Asterisk
> http.c: HTTP session count exceeded 100 sessions.
> We make the adjustment in the http.conf file
> sessionlimit = 500
> The error of SYN flooding persists, I clarify that the Firewall has blocked the external traffic so the connections to port 8089 are legitimate.
> I can see that when executing the command.
> netstat -anp | awk '{print $ 4 "" $ 6}' | grep '8089 ESTABLISHED' | wc -l
> Shows values ​​below the 100 allowed sessions.
> Verifying with netstat we find that there are more than 100 sessions in CLOSE_WAIT status, and do not finish the process by gluing the sessions until the SYN flooding error in Kernel happens.
> What is the recommended configuration so that this situation does not occur as it becomes recurrent.
> I appreciate any configuration parameter that helps us improve this instability in the system, thanks.



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