[asterisk-bugs] [JIRA] (ASTERISK-30119) res_pjsip_sdp_rtp: Timeout doesn't handle bundled streams

[Torbjörn Abrahamsson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30119?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259935#comment-259935 ] 

Torbjörn Abrahamsson commented on ASTERISK-30119:
-------------------------------------------------

Could it possibly as easy as changing: 

session_media->type == AST_MEDIA_TYPE_VIDEO

to: 

(session_media->type == AST_MEDIA_TYPE_VIDEO && !session_media->bundled)

in the second of the two changes in the patch mentioned in the description?


> res_pjsip_sdp_rtp: Timeout doesn't handle bundled streams
> ---------------------------------------------------------
>
>                 Key: ASTERISK-30119
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30119
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp
>    Affects Versions: 16.24.0, 16.25.0, 16.26.0, 16.27.0
>            Reporter: Joe Searle
>            Severity: Major
>              Labels: webrtc
>         Attachments: endpoint.txt, video_recvonly.pcap, video_sendrecv.pcap
>
>
> Since upgrading from 16.22.0 to 16.27.0 when making outbound calls from my sip client (webrtc) I have been noticing Asterisk has been dropping the calls slightly after the 60 second mark with the notice
> {quote}[Jun 30 13:05:34] NOTICE[20479]: res_pjsip_sdp_rtp.c:149 rtp_check_timeout: Disconnecting channel 'PJSIP/softphone-web-6-00000002' for lack of video RTP activity in 65 seconds{quote}
> I used git bisect to track down the issue to this commit
> {quote}14156f9827eb0caa4e3b64bafce318e475a12b68{quote}
> The related jira issue is ASTERISK-28890
> Prior to this change I wouldn't have any issue with Asterisk dropping the calls.
> I thought that the problem was that the INVITE sent from the endpoint to Asterisk was set to *recvonly* on the video stream and that Asterisk wasn't checking whether it should be receiving the video RTP before it adds the applies the rtp_timeout to it. This does not seem to be the case however because I tested enabling video in the softclient before making another attempt so it was set to *sendrecv* but I still have the problem with video being sent to Asterisk.
> Reading the linked jira issue the change relates to *rtp_keepalive* which I don't have enabled on the endpoints.
> I will attach the endpoints settings and a pcap once I figure out how.
> The only work around I have found so far is to disable rtp_timeout on the endpoints which isn't ideal.



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