[asterisk-bugs] [JIRA] (ASTERISK-17744) RTP timestamp skewed after call transfer or call unhold

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-17744?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan closed ASTERISK-17744.
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    Resolution: Won't Fix
    
> RTP timestamp skewed after call transfer or call unhold
> -------------------------------------------------------
>
>                 Key: ASTERISK-17744
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-17744
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Core/RTP
>    Affects Versions: 1.8.3
>            Reporter: xxot-alex
>            Severity: Minor
>         Attachments: timestamp_issue_bug.pcap
>
>
> During incoming call from outside on asterisk after call transfer (which is performed on asterisk) to another extension remote party lost incoming audio channel (one way audio). We find out that the reason is connected with RTP timestamp which is jumped to huge abnormal values after transfer or putting call on hold. This issue was described here
> https://issues.asterisk.org/view.php?id=11491
> and I believe here: https://issues.asterisk.org/view.php?id=17007
> But there was bug in version 1.4. In 1.8.3.3 we have the same: timestamps skewed to crazy values. In documentation to RTP there is an information that RTP Marker should be set if RTP source is changed. It is really set here, but many phones use timestamps for counting jitter and don't pay attention on RTP Marker. This is documented bug in phones Cisco 7960 and it is still not fixed. Is there any chance to fix this issue in asterisk and keep RTP timestamps stable even after call transfer?

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