[asterisk-bugs] [JIRA] (ASTERISK-27685) PJSIP outbound-publish, ETag and Event: dialog

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27685?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242243#comment-242243 ] 

Asterisk Team commented on ASTERISK-27685:
------------------------------------------

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].

> PJSIP outbound-publish, ETag and Event: dialog
> ----------------------------------------------
>
>                 Key: ASTERISK-27685
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27685
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 15.2.1
>            Reporter: Cyrille Demaret
>              Labels: pjsip
>
> Asterisk is sending a PUBLISH with a "early" state with a SIP-If-Match for an already terminated dialog. According to the RFC 4235, this shouldn't happens.
> Asterisk send :
> PUBLISH sip:201 at 192.168.100.37 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj1794e605-4e31-4b85-a5c8-6e2d6b04896c
> From: <sip:201 at mydomain.com>;tag=5d7d5883-8dbe-4039-a58a-e6a7f97211a9
> To: <sip:201 at mydomain.com>
> Call-ID: 23b4c0df-ccf6-4325-912a-7396a0d169a4
> CSeq: 39221 PUBLISH
> Event: dialog
> Expires: 180
> Max-Forwards: 70
> User-Agent: Asterisk PBX 15.2.1
> Content-Type: application/dialog-info+xml
> Content-Length:   247
> <?xml version="1.0" encoding="UTF-8"?>
> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="15" state="full" entity="sip:201 at mydomain.com">
>  <dialog id="201" direction="recipient">
>   <state>early</state>
>  </dialog>
> </dialog-info>
> The presence server replies :
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj1794e605-4e31-4b85-a5c8-6e2d6b04896c;received=192.168.100.37
> From: <sip:201 at mydomain.com>;tag=5d7d5883-8dbe-4039-a58a-e6a7f97211a9
> To: <sip:201 at mydomain.com>;tag=b596189c6de9c38f624fd84638f43be6-dec6
> Call-ID: 23b4c0df-ccf6-4325-912a-7396a0d169a4
> CSeq: 39221 PUBLISH
> Expires: 170
> SIP-ETag: a.1518814633.30600.10.0
> Server: kamailio (5.0.5 (x86_64/linux))
> Content-Length: 0
> When the call is done, Asterisk send another PUBLISH telling that the call is terminated :
> PUBLISH sip:201 at 192.168.100.37 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj537dc03b-eb54-4e99-b221-beb7352be34f
> From: <sip:201 at mydomain.com>;tag=71c848f9-a9e7-4e63-a990-bc604fdcada5
> To: <sip:201 at mydomain.com>
> Call-ID: 23b4c0df-ccf6-4325-912a-7396a0d169a4
> CSeq: 39222 PUBLISH
> Event: dialog
> SIP-If-Match: a.1518814633.30600.10.0
> Expires: 180
> Max-Forwards: 70
> User-Agent: Asterisk PBX 15.2.1
> Content-Type: application/dialog-info+xml
> Content-Length:   230
> <?xml version="1.0" encoding="UTF-8"?>
> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="17" state="full" entity="sip:201 at mydomain.com">
>  <dialog id="201">
>   <state>terminated</state>
>  </dialog>
> </dialog-info>
> The presence server replies :
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj537dc03b-eb54-4e99-b221-beb7352be34f;received=192.168.100.37
> From: <sip:201 at mydomain.com>;tag=71c848f9-a9e7-4e63-a990-bc604fdcada5
> To: <sip:201 at mydomain.com>;tag=b596189c6de9c38f624fd84638f43be6-9a38
> Call-ID: 23b4c0df-ccf6-4325-912a-7396a0d169a4
> CSeq: 39222 PUBLISH
> Expires: 170
> SIP-ETag: a.1518814633.30590.9.1
> Server: kamailio (5.0.5 (x86_64/linux))
> Content-Length: 0
> If a new call is made before the expiration, Asterisk reuse the same ETag:
> PUBLISH sip:201 at 192.168.100.37 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj95f84975-2ec7-4487-a038-e93098b8373d
> From: <sip:201 at mydomain.com>;tag=58c0292a-56d1-4dde-9824-3fa9f2153863
> To: <sip:201 at mydomain.com>
> Call-ID: 23b4c0df-ccf6-4325-912a-7396a0d169a4
> CSeq: 39223 PUBLISH
> Event: dialog
> SIP-If-Match: a.1518814633.30590.9.1
> Expires: 180
> Max-Forwards: 70
> User-Agent: Asterisk PBX 15.2.1
> Content-Type: application/dialog-info+xml
> Content-Length:   247
> <?xml version="1.0" encoding="UTF-8"?>
> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="19" state="full" entity="sip:201 at mydomain.com">
>  <dialog id="201" direction="recipient">
>   <state>early</state>
>  </dialog>
> </dialog-info>



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