VoIP - fixing voice quality.

Delayed speech and dead spots (usually called choppy voice) are usually associated with latency, jitter, packet loss and packet discards.  These issues are related to conditions that can exist between the two endpoints of a voice transmission.  A VoIP connection has these transmissions occurring inbound and outbound, as a conversation is carried in both directions and these transmissions can take different paths across the Internet.  Typical broadband connections are asynchronous, meaning that they have more download than upload.  This can come into play, as the outbound audio (the upload) can experience the need for more bandwidth, especially if other applications are using some of that bandwidth.  In many cases, these quality issues can result from a few identifiable causes, some of which could be congested points along the path that the voice has taken.

 

Delayed Speech and Long dead spots:

This can be the result of latency or delays on the network or Internet.  In some cases, you could experience brief periods where this issue could occur, but if it lasts for more than a day or two, then there may be a problem on the route that your VoIP signals (RTP) are traveling.  (Latency is a common issue with satellite connections, so if your broadband connection is from satellite, then this issue could be related to the actual technology itself.  In most cases a satellite connection is not adequate for VoIP, even on compressed codecs (although technology may end up conquering this dilemma someday), because the transmission is burstable which creates significant jitter.  This is addition to the latency that these connections create.) 

If your router has QoS settings, then enable QoS for your VoIP connection, typically RTP and SIP.  This will prevent applications that are running on your LAN from "stepping on" your VoIP (voice) connection.  We have a page that shows how to enable QoS for VoIP on a BEFSR41 router, (only the later models have this feature). [read more]

The best test for unusual latency is to measure the ping times between the actual hops (or routers) that the RTP (voice) signal is taking.  In the case of VoIP, these transmissions are usually UDP and to reproduce the exact paths can be difficult.  The next best thing would be to test using a ping test.  One utility that not only shows the individual routers, but also shows latency and packet loss, is PingPlotter, which can be used to troubleshoot your VoIP connection over the Internet.  If significant latency from your location outward occurs, this would be a significant indication that this latency is causing your voice issues.  There is also a very good VoIP test designed to give you results designed to measure specifics that would degrade a VoIP connection.  After connecting with the site you can choose a destination city.  A simulated call is then measured and you will be presented with a graph indicating a MOS Mean Opinion Score.  Several factors go into the calculation for a MOS score, some of which are latency, jitter, time to setup the call and the codec that is chosen.  You will find detailed information about their test at the site.

In many cases, latency and packet loss may be the result of issues that are occurring somewhere from the modem out, and the best place to start is right with your particular connection. This would especially be true if day after day you are experiencing the same issue over a period of time.  If these problems were somewhere out further on the network, then the chances are that many people would be experiencing them and the ISP would be trying to correct it, to prevent a prolonged issue.  As hopefully they would not want to have customers upset at their service.  (This is not to say that this does not occur, but rather to troubleshoot close in first.)

It might be the result of a signal issue at your modem or packet loss locally sometimes being caused by a splitter or faulty connector or cable line.  Check out our page on WAN Connectivity and the Internet for advice.