[asterisk-bugs] [JIRA] (ASTERISK-27685) res_pjsip_dialog_info_body_generator: Dialog id is reused after transitioning to terminated

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27685?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242524#comment-242524 ] 

Joshua Colp commented on ASTERISK-27685:
----------------------------------------

The underlying problem here is that dialog-info+xml is used to convey dialog state information, while we fake it for use for presence. In the case of dialogs you can't transition out of terminated. This seems tolerated by everything except for Kamailio. The fix would be to change the dialog id when we transition into terminated so that the next update has a unique dialog.

> res_pjsip_dialog_info_body_generator: Dialog id is reused after transitioning to terminated
> -------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27685
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27685
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_dialog_info_body_generator
>    Affects Versions: 15.2.1
>            Reporter: Cyrille Demaret
>            Severity: Minor
>              Labels: pjsip
>         Attachments: Publish.txt
>
>
> 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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