[asterisk-bugs] [JIRA] (ASTERISK-23142) Asterisk 11 not change SSRC on call transfer if marker is set and timestamp jumps

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23142?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214057#comment-214057 ] 

Matt Jordan commented on ASTERISK-23142:
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I don't believe this is a bug.

Looking at SSRC 0x445FBF7D, at at 6.039734 a period of time occurs where Asterisk is not sending RTP to the destination. At 6.244873, we resume sending.

The timestamp jump for the SSRC in question reflects reality: during the time period where the bridge is broken and the transfer is being performed, there is a space in time where no RTP is sent from Asterisk to the other source. When a bridge is re-established, the delta in the timestamp is reflective of the last RTP that was sent. So the combination of SSRC, timestamp, and sequence number is correct for that stream.

>From Asterisk's perspective, the RTP flow from itself to the UA that it established a session with hasn't changed. The fact that it is bridged with another source is immaterial: Asterisk is a B2BUA and the RTP stream established with the other UA in the bridge is an entirely different RTP stream. That shouldn't result in another SSRC/timestamp/seq no for the stream for the other UA, which is unbroken.


                
> Asterisk 11 not change SSRC on call transfer if marker is set and timestamp jumps
> ---------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23142
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23142
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.6.0, 11.7.0
>            Reporter: Filip Frank
>         Attachments: iptel207setting.txt, 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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