VoIP call quality
So we did a bit of testing last week. Spoke with Sean over VoIP with my ATCOM-320 SIP/IAX2 phone calling his home PSTN phone. Voice quality seemed pretty good using the 711a codec (8KB/s I believe) but there was some real stuttering going on. Most of the time it was smooth but there was enough stuttering to get a bit annoying. I think once I can figure out what is causing that, we will have good call quality. Unfortunately I'm not a network guru so where the problem lies should be interesting to find out.
What we know of the network is:
- Server end currently is 3Mbit DSL on a Linksys WRT54G
- My home is 4Mbit DSL on a Linksys WRT54G
Placing one call between these two via the 711a codec @ 8KB/s shouldn't be a problem. Especially since the two are on the same ISP which via a ping to the PSTN provider is 35ms. Is this acceptable for VoIP? I don't know but it doesn't sound like a bad ping response to me.
Testing the ATCOM-320 and our Softphone software built into Nexxia, results in the same kind of stuttery/packet missing kind of sound. So it can't be the phone nor the software.
Will have to devise a way to do some network analysis.
More to come soon!
What we know of the network is:
- Server end currently is 3Mbit DSL on a Linksys WRT54G
- My home is 4Mbit DSL on a Linksys WRT54G
Placing one call between these two via the 711a codec @ 8KB/s shouldn't be a problem. Especially since the two are on the same ISP which via a ping to the PSTN provider is 35ms. Is this acceptable for VoIP? I don't know but it doesn't sound like a bad ping response to me.
Testing the ATCOM-320 and our Softphone software built into Nexxia, results in the same kind of stuttery/packet missing kind of sound. So it can't be the phone nor the software.
Will have to devise a way to do some network analysis.
More to come soon!
posted by Simon at
9:49 AM
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