[asterisk-bugs] [JIRA] (ASTERISK-25840) Asterisk 13.7.0 unable to send INVITEs to jsSIP (WebRTC) peer connected over WSS

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25840?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=229866#comment-229866 ] 

Asterisk Team commented on ASTERISK-25840:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> Asterisk 13.7.0 unable to send INVITEs to jsSIP (WebRTC) peer connected over WSS
> --------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25840
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25840
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/WebSocket
>    Affects Versions: 13.7.0
>         Environment: Server: Debian wheezy; 
> Client: jsSIP 0.7.17; Chrome/Firefox: latest
>            Reporter: Kirill Marchuk
>
> Long story is here: https://groups.google.com/d/msg/jssip/1r6L2R0i1vk/fBrU7HAHBgAJ
> Long story short: 
> as we connect jsSIP client to Asterisk over WSS, we can call and register, but we don't receive calls. Over WS we receive calls even with jsSIP; with sipML5 we receive calls even with WSS.
> I've posted this on jsSIP discussion group and their core maintainer (Inaki Baz Castillo) explained me that this is an Asterisk issue (see his reply to my message at the link above)
> I can confirm that "tcpdump port 8089" does NOT display anything when I run "channel originate SIP/<peer> application Playback hello-world", although I can see this in CLI: 
> Reliably Transmitting (NAT) to 188.133.x.z:51147:
> INVITE sip:2lansdgc at 6o5gvecvk9qu.invalid;transport=ws SIP/2.0
> (I was pretty shocked that such a message in CLI does not mean that packet was actually sent, so this is a bug per se, IMHO)
> afterwards it does some attempts of retransmitting this packet, none of these appear in "tcpdump port 8089" output
> Can you confirm this is a bug or should I provide any additional information ?



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