We're running a VOIP system over Comcast. The problems are:

  1. Some calls don't complete (both incoming and outgoing)
  2. Some phone calls get terminated in the middle of the call.

I want to run some kind of throughput test on the Comcast service (which could test the throughput several times per second) to ensure that there is enough bandwidth available for the phonecalls. Which free or trial software or applications would you recommend?


For the most part, VoIP is not bandwidth-intensive. It is, however, quite sensitive to variable latency. In your testing, you need to test both aspects. I'd recommend you look at setting up an iperf server on one end of your connection. You'll then connect to that from a client on the other end. Iperf will be able to not only report on the bandwidth available, but also on the jitter of the connection, which is quite useful when troubleshooting VoIP issues.

Additionally, you may want to look at doing some real SIP tests using SIPp. You'll be able to send incrementally increasing numbers of calls through and find where things start breaking.

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From what I have learned spikes in latency making VoIP and other applications unusable in cable modem systems stems from the buffer bloat issues described a few years back. Specifically, cable modem front ends (UBRs) that may hang onto a packet for many seconds while waiting for an open upstream slot to send it on. I am sure there are many much more expert than I on this list but this explanation fits the problems I too have been struggling with over the past few years. One of the partial solutions is to make sure you never exceed the available upstream bandwidth. This can easily be done with a real router or any of the open source routers out there (I use vyatta but pfsense and others are fine as well as a used cisco off ebay). From the links found from the below link, the idea is to use traffic shaping on outgoing packets to match available upstream bandwidth so that the UBR never has to buffer an outgoing packet and hang onto it "forever". However, the reason I have found this to be only a partial solution is that the "available" bandwidth varies with time of day/number of users/etc... So even on my 20mbps link, the available bw sometimes drops to less than a megabit at times. To limit my upstream to 1mbps was just too bitter a pill to swallow when I normally have 20. So I just bought a 384kbps DSL line for exclusive use by my VoIP PBX and all the latency problems went away. Variation to my SIP provider over DSL is never more than 50ms (unlike the >8000ms I had over cable). This all being said, I would love to find a more elegant/cost-effective approach to this. Comcast clearly has a way to prioritize IP traffic for their own VoIP offerings.


for discussion and some things like backing off upload speeds that might improve things.

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iperf is a useful bandwidth testing tool. You run it in server mode on a remote server, then run a client to connect to it. This is a Linux/BSD tool, so you may need a LiveCD.

You might be interested to know that we routinely have trouble as well, due to large latency spikes within Comcast's network.

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  • Thanks for your answers. draeath, how did you solve your problems? Did you switch to a T1 or did you make some changes to Comcast? – Charles Apr 22 '11 at 23:07
  • Hi Charles. We're still having the problems, after some workaround to make them sting less. We have our own fiber link scheduled for installation as a permanent solution. One particular note was that we had to tweak the "qualify" sip setting for some of the phones, as they would unregister if they were polled while our ISP was waving 'issues' - we have not found any way to help with the call quality or drops, though. – draeath Apr 27 '11 at 16:48

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