[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=214075#comment-214075 ] 

Matt Jordan commented on ASTERISK-23142:
----------------------------------------

Looking at the two packets I referred to:

|| Time || Source || Destination || SSRC || Seq || Time ||
| 6.039734 | 10.76.17.2 | 10.76.17.130 | 0x445FBF7D | 10492 | 23720 |
| 6.244873 | 10.76.17.2 | 10.76.17.130 | 0x445FBF7D | 10493 | 25360 |

So there isn't a jump here.

I did, however, take another look at the RTP stream using Wireshark's RTP analyzer, and it did point out the packet that you were referring to - which actually occurs at 7.374434. At that point in time, the timestamp does jump from 33680 to 1038379896, which definitely isn't correct. So the bug is at that point, and not where I originally looked.

As an aside, with large pcaps, it does help us if you point out where you think the problem is, as opposed to letting us dig through it and try to find it ourselves (or, in this case, miss it the first time around...)


                
> 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.

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira