There are several issues which can affect quality of On-net calls (SPANtalk to SPANtalk) including:

  • Low bandwidth internet connection. A 256/64 basic ADSL should be sufficient when G.729 codec is used, but if G.711 is used for a particular call then quality will suffer.
  • Poor quality phone line (for ADSL) can mean that the actual throughput is noticeably less than the synchronisation speed.
  • Modem/router configuration. Older routers generally require ports opened for VoIP (and the RTP ports depend of the VoIP device being used).  Also, some routers block outgoing ports as well as incoming.
  • Using multiple VoIP devices can be problematical, especially if router ports had to be forwarded/opened.
  • Other internet traffic which is trying to share the internet connection.  P2P file sharing problems are most likely to affect VoIP due to large number of connections, and longer timeframe for the download/upload.  QoS can help here.
  • Faults and congestion over the internet happen from time to time; but since VoIP is time-critical, any problems are more noticeable.
  • Calls to some countries with highly regulated internet communication , due to (a) inadequate bandwidth for the demand, and (b) government censorship and controls.

Off-net calls (SPANtalk to PSTN) can be affected by all the above (for the portion to the SPANtalk PSTN gateway) plus:

  • The evening is peak-time for SPANtalk, and it appears that there may not always be enough PSTN lines for the number of calls

Unfortunately it is not easy to check the quality of an on-net call.  You would need to determine the public IP Address of the other party in order to test using ping and tracert.  The higher the ping times, the more chance that packets will be lost.

Please consider the following before submitting a Helpdesk Ticket:

  1. What are the symptoms?  Are you able to hear the person at the other end OK?  Can they hear you OK?  Is the problem too much echo, choppy sound, or something else?  Does this affect only calls to/from certain people, calls to all PSTN numbers, or calls to other VoIP users as well?
  2. Is QoS enabled?  If you are using a Vigor 2100V or 2900V with an external modem, you need to tell it how much upstream bandwidth is actually available to your ISP, so it knows when to start applying QoS. Please see “What Bandwidth should I set for my broadband router?”
  3. How much upstream bandwidth is available, and is it also being used by other applications (such as P2P file sharing)?  If you turn off the other applications, does the call quality improve?
  4. What default codec are you using?  Note that not all devices support all codecs; for example we recommend using G.729 because it gives excellent quality using only 8Kbps – but G.729 uses patented code which requires licence payments, so free VoIP softphones (e.g. X-Lite and Asterisk) often don’t support it.  Fortunately the two SIP devices negotiate which codec to use on a call-by-call basis.

During a call please look at the router’s VoIP > Voice Call Status to see what codec is actually being used for those calls with poor call quality.

If you have a basic 256/64 ADSL connection, any calls using G.711 codec will suffer due to trying to send 64Kbps (plus some overhead) up a pipe which only takes 64Kbps (less some overhead).  Other internet activity at the same time will only aggravate this situation.