Eliminate double transcoding G.711 to RTAudio RRS feed

  • Question

  • Currently in OCS 2007 (not R2), when a call comes in to the mediation server and is diverted back out again (e.g. due to call forwarding or sim ring set on the OCS client to a PSTN number) the call appears to be trancoded from G.711 to RTAudio and then back to G.711 again leading to a poorer call quality than if it would have remained in G.711.

    Is this quality issue addressed in OCS 2007 R2? Either by a higher bitrate transcoder or keeping the call as G.711?
    Thursday, May 7, 2009 9:01 PM

All replies

  • i wrote a blog post about this a day ago, and maybe it answers your question a bit.

    I found out today that the codec being used when making a PSTN call is now in some special conditions G711

    G711 may be used for calls to the PSTN depending on a number of network characteristics:

    • Round trip delay of less than 20ms
    • Call originates from the LAN

    So why change from RTAudio since we have been told that it is so superior? Well apparently RTAudio uses more CPU and is a wide band codec witch can not be used in a PSTN call anyway.

    But if Communicator detects changes in the network characteristics, such as an increase in the round trip delay, the codec will automatically switch to RTAudio in order to ensure the call continues without issues.

    Tommy Clarke | http://www.cinline.se
    Thursday, August 13, 2009 8:38 PM
  • Not sure if there is any change in behaviour when a call comes into the Mediation Server and forks to PSTN again

    Be aware that communicator itself switches to Narrowband when doing PSTN calls so there is no wasted overhead of CPU bandwidth

    RT-Audio has 9 different bit rate possibilities and scales dynamically
    More information about the codec can be found here

    - Belgian Unified Communications Community : http://www.pro-exchange.be -
    Friday, August 14, 2009 12:34 PM