Voxalot / SIP Broker Support Forums - View Single Post - Poor quality of voice comments and questions

loraolo · 11-19-2007, 11:53 AM

I'm experiencing poor voice quality in some pick time mainly due to fluctuation of internet speed (i.e jitter, mos, ...).
I also found that some problems related to one way conversation (i.e able to hear voice but no voice out) could be strictly depending on the codec used.
(i.e. I tried to set GSM as first preferred codec for voxalot-betamax provider. The problem disappeared as I set as preferred codec G729, PCMA (alaw), ...)

Also calls voxalot to voxalot are affected of the same problem. Voxalot can decide to choose iLBC or G711 codec but due to delay variations on internet speed the conversation could be interrupted after a while or simply lose packets. I connected a Grandstream ATA HT488 with a bridge PSTN to VOIP connection for incoming calls. Sometime the call is broken after a while and other times is perfect.

How can I simulate a SIP connection to Voxalot and monitor the result over time in order to troubleshoot connections problem? Do you provide a sort of client-server tool able to do that?

Thanks
Paolo

11-19-2007, 11:53 AM	#1
loraolo Member Join Date: Jul 2007 Posts: 36 Thanks: 0 Thanked 6 Times in 3 Posts	Poor quality of voice comments and questions I'm experiencing poor voice quality in some pick time mainly due to fluctuation of internet speed (i.e jitter, mos, ...). I also found that some problems related to one way conversation (i.e able to hear voice but no voice out) could be strictly depending on the codec used. (i.e. I tried to set GSM as first preferred codec for voxalot-betamax provider. The problem disappeared as I set as preferred codec G729, PCMA (alaw), ...) Also calls voxalot to voxalot are affected of the same problem. Voxalot can decide to choose iLBC or G711 codec but due to delay variations on internet speed the conversation could be interrupted after a while or simply lose packets. I connected a Grandstream ATA HT488 with a bridge PSTN to VOIP connection for incoming calls. Sometime the call is broken after a while and other times is perfect. How can I simulate a SIP connection to Voxalot and monitor the result over time in order to troubleshoot connections problem? Do you provide a sort of client-server tool able to do that? Thanks Paolo