The sound quality that I get for “real-world” SIP calls going out over the WAN is very poor, but if I make a SIP call from a client application (Xten Lite) on my actual Asterisk same server, the sound quality is perfect.
I pinged the gateway of my VoIP provider and I’m consistenly getting 75-80 ms delay on the reply back to my Asterisk server. Obviously, when I ping my own Asterisk server from the same machine, I get much less delay (under 1 ms).
Does anyone know what the delay threshold is before call quality signficantly suffers? I would imagine that 75 - 80 ms is way too high to expect good results Also, I’m using a DSL connection as my access circuit (approx. 1.3 Mbps down/384 up). Might I need to switch to Cable modem?
You should try other proxies of your provider, they should have more than one proxy except if its just a random person hosting an * box right in his bedroom.
as long as :
a) his provider supports IAX2
b) his provider supports iLBC
c) he is happy to introduce the translation time for iLBC (~13ms)
d) he is happy to introduce the processor cost for iLBC
Thanks for all of your help. I have finally found the problem and it’s with my VoIP provider. I set up a free GIZMO account and registered my Asterisk server as a client so that I could call in and test sound quality going over another VoIP provider’s network. Sure enough, the sound quality was clear and undistorted. I had my VoIP provider send my calls through another gateway and I still got the same result.
Now I need to find another VoIP provider who can provide:
Good sound quality on calls.
Flat rate for unlimited inbound calls.
Multiple channels that can be shared across a range of DIDs to allow many simultaneous inbound calls.
DIDs from a variety of US cities (and preferably international numbers too).
Any recommendations would be greatly appreciated. My current VoIP provider offers all of the above EXCEPT the good sound quality. Otherwise, I’d stay with them.