[asterisk-bugs] [JIRA] (ASTERISK-23142) Large timestamp skew in RTP stream during blind transfer

[Filip Frank (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23142?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Filip Frank updated ASTERISK-23142:
-----------------------------------

    Attachment: asterisk_rtp_large_transfer_skew.patch

I add my patch, we use it on about 70 of our customers PBX, without any problem. But we not using jitterbuffer for SIP, because it causes more errors in audio especially with transfers. 

> Large timestamp skew in RTP stream during blind transfer
> --------------------------------------------------------
>
>                 Key: ASTERISK-23142
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23142
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.6.0, 11.7.0
>            Reporter: Filip Frank
>         Attachments: asterisk_rtp_large_transfer_skew.patch, iptel207setting.txt, rtp_instance_debug.patch, rtp_timestamp_jump.pcap
>
>
> I have problem with asterisk 11, I call from phone A (ip 10.76.17.130 - iptel207) to phone B(iptel106). After answer the call on B, i use blind transfer asterisk feature to phone C(iptel500). Problem is after bridge with A and C, the timestamps in RTP from asterisk to phone A jumps. In first RTP packet with jumped timestamp the market bit is set, but SSRC is not changed. This is test situation, but in real i have a problem if A is Ericsson IMS(operator O2). They drop audio from our asterisk after timestamp jump and A dont hear  C. I my attached pcap trace bad RTP is with SSRC 0x445FBF7D. I see in wireshark big skew too....
> I found this in RTP RFC3550:
> " All packets from a synchronization source form part of the same
>       timing and sequence number space, so a receiver groups packets by
>       synchronization source for playback."
> Then I think timestamps cant jump without SSRC change. We not use direct media. All RTPs flow over asterisk.



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