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  1. FreeSWITCH
  2. FS-9826

one way audio when SSRC changes while jitter buffer is paused

    Details

    • Type: Bug
    • Status: Resolved
    • Priority: Minor
    • Resolution: Fixed
    • Affects Version/s: None
    • Fix Version/s: 1.6, 1.8
    • Component/s: freeswitch-core
    • Labels:
      None
    • Environment:
      CentOS 6.6 x86_64
    • CPU Architecture:
      x86-64
    • Kernel:
      Linux
    • Userland:
      GNU/Linux
    • Distribution:
      CentOS
    • Distribution Version:
      CentOS 6
    • Compiler:
      gcc
    • FreeSWITCH GIT Revision:
      c0e3224f6ecf8f86db6babf58f9f8d3813266fad
    • GIT Master Revision hash::
      c0e3224f6ecf8f86db6babf58f9f8d3813266fad
    • FSS Support Agreement Customer Number and Company name:
      Grasshopper

      Description

      This scenario happens when I have a bridged call with paused jitter buffers in both call legs. One of the call legs places the call on hold with SIP re-INVITE, then takes the call off hold with another SIP re-INVITE. During this time, the SSRC for the incoming audio stream has changed for each re-INVITE.

      I can overcome this issue by manually resuming then pausing the jitter buffer or by making a change to switch_rtp.c to reset the jitter buffer when SSRC changes regardless of jitter buffer paused state.

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            People

            • Assignee:
              mikej Mike Jerris
              Reporter:
              crienzo Christopher Rienzo
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              Dates

              • Created:
                Updated:
                Resolved: