you can get a pcap of the rtp from the call and analyze it. also you could compare to the quality of using the gmail applet. I suspect it just is bad and they should consider using FS on the other side to convert it to sip =D On Wed, Sep 8, 2010 at 11:26 PM, Larry Matter <cm-freeswitch@matter.net> wrote:
I was very excited to see that freeswitch can make calls through google voice (or is gmail?). I got it working without too much fuss thanks to the wiki. But the sound quality compared to my normal sip trunks is not very good. I'm wondering if I'm doing something wrong or do others have this "get what you pay for" experience with it.
When I made a call the experience was a very significant delay in the audio (like a few seconds). I would hear a question, and answer it, and then hear the other person repeat the question as if I didn't hear it. There was also significant "crackle" as if there was a loose wire somewhere. I called back using my regular sip trunk and it was fine.
I run freeswitch on an underutilized x86_64 box running Centos 5.5, zippy broadband connection. It is behind a NAT device, and my Cisco 7940 is behind a different NAT device. I'm a novice when it comes to the details of VOIP, but of course love to learn. I would love some suggestions on how to investigate this.
Thanks! Larry Matter
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