Call Quality Troubleshooting
In the case that you are experiencing bad call quality please consider the following suggestions.
1. Please check your ping time to the sip/iax trunk provider that you use. The basic rule is that you may want to connect to the closest one to your server. Fortunately, most sip trunk provider has multiple locations which is very close to our Seattle or New Jersey server. Please do your investigation and find your best path to your sip trunk location and use it.
2. Unlike your Local Area Network, internet will have jitter (small or large depending on quality of your ISP). This is the worst issue with VOIP. However, please do not be discourage. Most voip phone and ATA device will have jitter buffer to compensate the issue. Some will have it by default and not telling you that it have one. Please investigate your phone for this feature. Your supplier and google is the best way to find this information. At the end of the day, please make sure that you have this feature enable on the phone. Asterisk SIP and IAX has jitter buffer setting. PLEASE DO NOT TURN THEM ON IN THE ASTERISK. They are for different purpose an may degrade you call quality.
3. Please, help your ISP, Internet, Your router, and us. Turn on your qos setting on your phone. Most phone can set tos falg. Set them correctly. Here is some info about tos. http://en.wikipedia.org/wiki/Type_of_Service. http://www.voip-info.org/wiki/view/DiffServ. Please make sure that your routers and phone will work together to prioritize your audio packet.
4. As the last step, after you have done the above and feel that the experience can be improved more, you may want to play around with different codec. I recommend g729. Please note , you may need to pay for additional license depending on your usage. If you need more info about g729 in the asterisk world, please refer to http://www.voip-info.org/wiki/view/Asterisk+G.729+Licensing . In my personal experience, using better codec yield the best bang for you effort.