[asterisk-bugs] [JIRA] (ASTERISK-26642) pjsip: Improper Handling of extra 200 OK from forked INVITE

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26642?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234072#comment-234072 ] 

Joshua Colp commented on ASTERISK-26642:
----------------------------------------

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

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

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> pjsip: Improper Handling of extra 200 OK from forked INVITE
> -----------------------------------------------------------
>
>                 Key: ASTERISK-26642
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26642
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Third-Party/pjproject
>    Affects Versions: 13.11.2
>            Reporter: Tim L
>            Severity: Minor
>
> I am using Kamailio as a SIP Proxy in front of multiple Asterisk servers. I am using 13.11.2 with pjsip. Users can register with Kamailio from multiple devices and Kamailio is performing parallel forking of INVITEs sent to those users. Kamailio will CANCEL all but the first INVITE that receives a 200 OK. However, if two devices respond with a 200 OK at the same time, Kamailio forwards both 200 OKs to Asterisk. I believe Kamailio is required to do this so that a UAC can send an ACK+BYE for all but one of the 200 OKs.
> What seems to be happening is that neither dialog is actually ended with a BYE, but the SDP answer from the second 200 OK is used, leaving the first one basically unusable. 
> Thank you for your support!



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