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

[FlashSystems (JIRA)]

FlashSystems created ASTERISK-28823:
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             Summary: 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


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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