[asterisk-bugs] [JIRA] (ASTERISK-28823) Updates for outgoing registrations not sent to the correct network address

[George Joseph (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28823?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

George Joseph updated ASTERISK-28823:
-------------------------------------

    Assignee: Unassigned  (was: FlashSystems)
      Status: Triage  (was: Waiting for Feedback)

> Updates for outgoing registrations not sent to the correct network address
> --------------------------------------------------------------------------
>
>                 Key: ASTERISK-28823
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28823
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 17.3.0
>         Environment: Linux on x86_64
>            Reporter: FlashSystems
>            Assignee: Unassigned
>         Attachments: debug_log, pjsip.conf
>
>
> If an outbound registration is performed the update of the registration (to keep it from timing out) is not done to the same network address as the initial registration. If more than one host is present in the SRV record, the outbound update of the registration is sometimes done to a different host.
> RFC 3261 (SIP: Session Initiation Protocol) states in Section 10.2.4 (Refreshing Bindings) that “(…)Registration refreshes SHOULD be sent to the same network address as the original registration, unless redirected.”.
> So subsequent registration requests should always reach the same host.
> The provider in question uses the following SRV-Records (redacted):
> {code}
> _sip._udp.phone.provider.example.   5452    IN      SRV     10 1 5060 phone-a.provider.example.
> _sip._udp.phone.provider.example.   5452    IN      SRV     10 1 5060 phone-b.provider.example.
> {code}
> As the following two `REGISTER` messages show the first one is sent to phone-a (IP: XXXX:7::10) the update for the registration is sent to phone-b (IP: XXXX:8::10):
> {code}
> [Apr 10 11:30:28] VERBOSE[26529] res_pjsip_logger.c: <--- Transmitting SIP request (616 bytes) to UDP:[XXXX:7::10]:5060 --->
> REGISTER sip:phone.mnet-voip.de SIP/2.0
> Via: SIP/2.0/UDP [YYYY::1]:5060;rport;branch=z9hG4bKPjdf58fb88-72ac-4494-9670-99ec1f0f8e9f
> From: <sip:+12345678 at phone.mnet-voip.de>;tag=74a4b55d-e91d-46bd-898c-02f2b0812802
> To: <sip:+12345678 at phone.mnet-voip.de>
> Call-ID: 7fda0159-790d-4c9d-a934-80e36a0d0d70
> CSeq: 26348 REGISTER
> Contact: <sip:+12345678@[YYYY::1]:5060;line=xrhrzif>
> Expires: 1200
> {code}
> {code}
> [Apr 10 11:50:18] VERBOSE[27575] res_pjsip_logger.c: <--- Transmitting SIP request (616 bytes) to UDP:[XXXX:8::10]:5060 --->
> REGISTER sip:phone.mnet-voip.de SIP/2.0
> Via: SIP/2.0/UDP [YYYY::1]:5060;rport;branch=z9hG4bKPj7c7043f0-57c3-40fb-90a5-77810229e0c1
> From: <sip:+12345678 at phone.mnet-voip.de>;tag=3e56a13f-a983-477e-be0b-1c9cb32e63b4
> To: <sip:+12345678 at phone.mnet-voip.de>
> Call-ID: 7fda0159-790d-4c9d-a934-80e36a0d0d70
> CSeq: 26350 REGISTER
> Contact: <sip:+12345678@[YYYY::1]:5060;line=xrhrzif>
> Expires: 1200
> {code}
> In my case the new registration to a different host leads to currently running calls being terminated.



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