[asterisk-bugs] [JIRA] (ASTERISK-23144) Asterisk 11 not change SSRC on call transfer if marker is set and timestamp jumps

[Matt Jordan (JIRA)]

Matt Jordan created ASTERISK-23144:
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             Summary: Asterisk 11 not change SSRC on call transfer if marker is set and timestamp jumps
                 Key: ASTERISK-23144
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23144
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_rtp_asterisk
    Affects Versions: 11.6.0, 11.7.0
            Reporter: Filip Frank


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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