Do you have coffee or a snack in hand? A bottled water perhaps? You might want to grab one because this article might just be an endurance test and you’ll want to stay hydrated! I promise you there is some good stuff here to help you understand some basic ISP and network problems that might be affecting your quality with VOIP/SIP applications. So stick with me if you can for the whole thing, or have this article up when you call your ISP, it might just help eliminate some of the confusion that ISP techs can sometimes offer.
What is VOIP and SIP? The short answer here is phone calls on the internet. Zipwire and Avaya are the applications VIPdesk Connect uses.
There are several factors that can affect VOIP/SIP. Great news is, it's not always about your speed!
The first major issue to consider is Jitter. Jitter is the amount of variation in latency/response time, in milliseconds. Reliable connections consistently report back the same latency over and over again. Lots of variation (or 'jitter') is an indication of problems.
Jitter issues show up with different symptoms, depending on the application you're using. Web browsing is pretty resistant to Jitter, but any kind of streaming media (voice, video, music) is going to suffer. Choppy calls, echo, and fading in and out are some examples of what Jitter can do with Bright Pattern.
A bad Jitter score is a symptom of other problems. It's an indicator that there might be something else wrong. Often, this 'something else' is bandwidth saturation (sometimes called congestion) - or not enough bandwidth to handle the traffic load.
Next is Bufferbloat. This is undesirable latency caused by routers and cable/DSL modems buffering more data than necessary. It occurs at any bottleneck in a network: the most common place is the connection between your router/modem and your ISP. It causes almost all VOIP services to degrade, and even stop working entirely.
Now let's talk about modems/gateways/routers. Not all are created equal. Some are simply not compatible for SIP. More commonly there is a setting, known as "SIP ALG" in modem/gateways as well as routers you might add to your home network. This setting was meant to fix NAT firewall issues and make calls pass through seamlessly and smoothly, but actually it has caused more harm with softphones than good. We generally like you to have it turned OFF at all times.
Hey since we're talking about changes your ISP tech can make, why not make it even EASIER to allow those calls to come through?
Ports and QOS, the technician might not be happy about doing this, as it takes a few extra minutes, but we need to open ports so please provide the tech with the following information:
Outbound firewall open TCP ports 80 and 8080 (HTTP)/443 and 8443 (HTTPS), 45025 and 45020 (HTTPS)
RTP audio stream, utilizes UDP, ports are dynamically allocated in the range 40,000 - 65,535
Outbound firewall UDP port 5080 open for SIP signaling
QoS router configurations recommended for UDP packets.
QoS packet prioritization policies are recommended for RTP and SIP
Hey we made it through to the end, thanks for sticking with me!
With this information you are well equipped to handle those not so comfortable calls with your ISP. You certainly don't have to fully understand this, they are paid to know and assist with these things, so don't be shy.
If you have any questions for our Help Desk Team, please do not hesitate to reach out, if your ISP has any questions, just write them down and pass them over to us in a ticket and we'll be happy to take a look!