First, ensure SIP ALG is not active in the router. If you are using a router that is set by default and cannot be changed, then the router must be changed. These are usually low-end routers that should be avoided like the plague.
Hosted VoIP is all about bandwidth and connection quality. It is not enough to ask your clients how good their internet is. Most users only browse or upload or download from their PC which is very forgivable. For instance, if it takes 3 seconds instead of 1 second for a page to pop up, no one notices this. But if it takes a voice packet 3 seconds to reach you instead of 40ms, it is usually dropped.
Pay close attention to bandwidth, we have several sites running a 16 by 1 DSL that handle the office data and VoIP easily while other sites with 25 x 10 will not handle both. It is all about how the customer uses their data, with more and more cloud services coming into play, the upload is just as important as the download. A call on VoIP has two modes, call setup, the bandwidth impact on this is negligible. Voice streaming (RTP), we do not compress voice as it gives a tinny sound to the voice, a voice stream in this case uses 64k for the voice and about 24k for the overhead, so a lot 88k per anticipated call. If the customer has an 880k upload, that does not mean you can do 10 calls, you will probably do 5 active calls at best, beyond that you will get increased latency.
If your host is, for example, demo.voice2net.ca, run winmtr on a laptop at the site to the host. You do not want to see anything above 60ms, your latency and jitter should be within 40ms. You can go above that but the customer will hear a lag in the voice. The real key is the latency and jitter should be consistent, not up and down, if it moves up and down, then you will hear pauses in the voice as the endpoint and the server realign to the moving target.
VoIP is all about communications between the host and the endpoints (IP phones). If you are using lamps and buttons on the phone, each time a device becomes active, messages are sent in the background to every device affected. A busy system with several phones and usage of busy lamps etc will cause a fairly large amount of background signalling. Some routers and firewalls handle this better than others. You should avoid home-style routers that were created for wifi usage in the home. Although they are good for their designed purposes, they do not function well in hosted VoIP environments. Our primary design is to install a quality router with no WIFI and use external access points for WIFI. That way each product is doing what they were both designed for.
When setting up a phone you have the option to choose UDP or TCP, Try to use TCP where possible, this only affects the SIP Messaging, not the RTP voice packets, the overhead increase is minimal and signaling improvement is substantial
The number of phones at an individual site should dictate whether you need separate networks or not. The router, data usage and bandwidth available and style of phone used will all impact this. If you are setting up 30 phones and most of them are 3 button phones, the signalling will be minimal and they will probably work with no problem on a 25x10 DSL mixed with the customers' internal network. However, 15 phones with extensive usage of busy lamps etc, could be a problem in the same scenario.
In all cases if you have access to fibre that is the best, second on the list is DSL, this is dependent on the speed available as previously discussed. Third, I place wireless ahead of Cable. Cogeco is not a stable network with numerous outages every month and Rogers is erratic. Some sites will work fine and others will not. Unfortunately, you do not know until you install them. In all cases, if using Cogeco or Rogers, you want their router in bridge mode and use your own quality router as the gateway.
The Switches you use or extremely important. If you use a Managed switch but do not manage it, you will likely have problems. We do not spend our lives in that area as we are prominently a VoIP Company. We have had several sites where a managed switch or firewall was installed that caused us untold grief until we manage to get the network people to work with us
Networks, if you manage the network for your customer, your life will be easy. Otherwise, we need to establish a relationship with your network company to ensure they take the VoIP Application seriously and realize there are companies besides Cisco in the world of networks.
It is time-consuming helping customers that do not understand they must have internet for the VoIP to work and the cables in the back of the phones have a specific spot they must be plugged into, and they do need to be connected to the network. This might seem like a joke, however, when we do 10 trouble calls in the field, I can assure you at least 7 of them were something similar to the above.
Traditionally a single phone. On these sites we do not apply all the rules we do to a busier business, so we quite often let them use whatever they have, but do advise them their home equipment could impact the voice over IP.
Voice2Net Corp.
246 King Street W, Prescott, On, K0E 1T0 Canada
Copyright © 2023 Voice2Net Corp - All Rights Reserved.
Powered by GoDaddy
We use cookies to analyze website traffic and optimize your website experience. By accepting our use of cookies, your data will be aggregated with all other user data.