Connect Network Configuration Guidelines
This is only a rough technical guide. If you have a large network setup, please get a qualified network technician.
Before you begin, you need to know:
The maximum number of concurrent calls you expect to have at each location.
Fonality uses two types of audio codecs G729 (default) and G711(backup). The bandwidth usage, assuming using Ethernet, is:
G729: ~ 33 kilo-bits-per-second (per call per direction)
G711 ~90 kilo-bits-per-second (per call per direction)
You should always plan with the busiest scenario in mind. For example, if you have 10 phones in total, expect, at some point all 10 phones will be in use. Using G711, you will need:
90 kbps * 10 = 900 kbps (kilo-bits per second) UPLOAD
AND
90 kbps * 10 = 900 kbps (kilo-bits per second) DOWNLOAD
How many phones do I plan to have at each location?
You shouldn’t use a residential router for enterprise environments. There is no exact technical cut-off line between using a residential vs. business router, a rough number would be <= 10 phones (which also assumes you have 10 computers at the same office). With under 10 phones, the Dlink DIR 655 should work great, if you have 20+ phones, strongly consider using something better. We have collected a list of routers we've used and their performance.
Fonality internal employees can also refer to this router listing guide
The brand and model of switch, router and modem I currently have at each location.
There are too many routers to give a to-do list for each type. When checking your existing networking equipment, please keep these in mind.
1.Is your IP phone connected this way?
a. IP Phone -> Router #1 -> Router #2 -> Internet
This is called double NAT. Fonality employees can refer to this article explaining why double NAT is a bad thing.
b.You will need to get your phone’s IP address and find the phone’s gateway. Log into the gateway (which would be router #1) and look for Router #1’s WAN IP address. If the WAN IP address listed is a local IP address, you have got, at least, a double NAT.
sc.When you have a double NAT, remove routers #2, (and #3, #4, etc.) You just need 1 router. If router #2 is actually a modem, you will need to contact your Internet Service Provider to bridge (disable the routing function) on that modem.
2.Check the SIP ALG (SIP Transformation, SIP Fixup) functions in your router.
a.Different routers (even different firmware versions of the same router) implements SIP ALG (a rule telling the router how to handle SIP traffic) differently. Most of the time, you will need to toggle the SIP settings on or off in the router to see which option is the most reliable.
3.Check SPI Firewall
a.If you experience call quality problems and your router has SPI enabled, try disabling it. SPI adds additional delay as it inspects each incoming audio packet. Would it make your network a little less secure? Yes. But that’s the trade off you have to make (or get an advanced router which can implement QoS).
b.QoS. Quality of Service
i.Your IP phones send audio on ports UDP 10000-20000.
ii.These audio packets are coming from the Fonality server. To get the IP address of the Fonality server your phones are using, find the registration domain name. It always is in the format: s<sid>.pbxtra.fonality.com. Configure the router to give priority to all inbound/outbound traffic between your network and this Fonality server.
4.*If you ONLY have 1 IP phone at this location, you can try port forwarding or DMZ modes on the router. You will need to port forward UDP 5060 and UDP 10000-20000. A good reference on how to do this is at: http://portforward.com/
What is the network latency at your location?
1.Latency
a.Latency is the time it takes for an audio packet to travel from your IP Phone to the other end. As a rule of thumb, <150ms of latency is tolerable.
b.To test latency, use a computer, on the same network as the phone, and ping Fonality's server at s<sid>.pbxtra.fonality.com. If you consistently see latency at over 300ms, it's not a good sign. When that happens, ping something else, such as google.com, if you also get 300ms or higher latency going to google, you should check with your ISP.
2.Jitter
Jitter is the variation in delay between audio packets. To give an extremly simplified example, you pick up the phone and say "Thank you". "Thank" is sent on Packet #1 and "you" is sent on packet #2. What if pakcet #2 arrives at the other end before packet #1? The other end would hear "you thank". There's not much you can do about jitter other than to make sure QoS is enabled on your routers. Fonality, by default, use jitter buffers to compensate for this problem. But there's a trade off, if the buffer is set too large, you will experience delay in audio (because the jitter buffer needs time to fill up).
3.Packet Loss
This is when stuff (audio) you sent gets dropped. This is the worst kind of problem and any reliable ISP should not have noticeable packet loss on their network.