[asterisk-bugs] [JIRA] (ASTERISK-25071) RFC3581 compliance

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25071?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226179#comment-226179 ] 

Joshua Colp commented on ASTERISK-25071:
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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



> RFC3581 compliance
> ------------------
>
>                 Key: ASTERISK-25071
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25071
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Registration
>    Affects Versions: 13.3.2
>         Environment: CentOS 6.5
>            Reporter: Oleg Rabinovich
>         Attachments: RFC3581.patch
>
>
> I am connecting to Asterisk via Kamailio SIP proxy (version 4.2.3). The client, MicroSIP-3.6.3 is behind NAT. Kamailio SIP proxy adds ";rport" to via headers in order to receive the traffic back from the server, according to RFC 3581. However, I fail to dial that client, as Asterisk dials directly to the client (public address) instead of my Kamailio box. This issue is similar to ASTERISK-7276, however, it's different in the sense that SIP request (INVITE) that is not routed correctly is a separate request from the one (REGISTER) that includes ";rport".
> I am attaching a patch that fixes the issue in my environment. Please, review it and let me know if this is a correct direction or am I missing something.



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