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David Vassallo's Blog

If at first you don't succeed; call it version 1.0

Improving call quality between AudioCodes Mediant Gateway and Asterisk PABX


Scenario: Asterisk PABX has been correctly and successfully set up to receive and send calls from / to an audiocodes mediant gateway. However, the call quality is unusable, with a lot of background noise and extremely choppy voice quality. The usual suspects, including those outlined below, have been ruled out:

  • Congestion issues: troubleshoot these issues by appling QoS, or in extreme cases of troubleshooting, simply connect the audiocodes gateway directly to the asterisk server
  • Asterisk Troubleshooting:
    • Do internal calls suffer from the same voice degradation? If so, then double check your extension settings, including SIP / IAX settings as well as the end point (softphone or hardphone) settings
    • Build an additional trunk between the Asterisk PABX and a different provider (for example a cheap, internet based SIP provider like mydivert.com). Do the voice degradation issues still occur? If so, double check your trunk settings, as well as general asterisk SIP settings. Note: if your voice quality degradation includes hearing a “clicking” or “ticking” noise in the background, read on.
  • AudioCodes Troubleshooting:
    • Does the call sound the busy tone only to certain destinations? If so, the destination may have limited capabilities. Remedy this by navigating to: VoIP > PSTN > Trunk Settings and setting
      • ISDN Transfer Capabilities : Speech
    • Do voice calls of every type suffer from voice issues? If so, double check your coders configuration section and try the following:
      • VoIP > Coders and Profiles > Coders: If your audiocodes and asterisk are both on the same LAN, decrease the packetization time to 10 (this will decrease the echo) and switch on silence suppression. Note: make sure these are the same coders being used by your trunk

If you still encounter the same background noise and choppy voice once you applied the above, only when using the AudioCodes, the problem is probably in the TDM settings between the AudioCodes and your provider. Remedy this by:

      • Ensure the AudioCodes configuration mode is set to full rather than basic
      • VoIP > TDM > TDM bus settings
      • Change PCM Law Select from MuLAW to ALaw
      • If the above still does not work, try toggling the setting TDM Bus Clock Source. Your mileage may vary

The above sorted out the background static and choppy voice, however we were still left with an irritating “ticking” or “clicking” noise in the background, roughly once every second when using the ALaw setting. To resolve this, we increased the Jitter Buffer in Asterisk. Using FreePBX:

  • Settings > Asterisk SIP settings > Jitter Buffer Settings
    • Jitter Buffer: Enabled
    • Force Jitter Buffer: No
    • Implementation : Adaptive
    • Jitter Buffer Size : 200 (jbmaxsize) and 1000 (jbresyncthreshold)

 

Sources and thanks:

 

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