Today, more and more companies build own VoIP system using IP telephony and unified communications. In turn, manufacturers also do not stand still and offer more and more IP telephony solutions. However, do not forget that the same SIP is not only a standard, but is a set of recommendations, where RFC 3261 is just a small part. When a manufacturer writes that it supports transferring a call or holding a call, this does not mean that this service will work with your IP-PBX. For example, your PBX supports a call transfer method according to RFC 3892 (REFERRED-BY), and the gateway supports the method according to RFC 3515 (REFER).
On the one hand, both options work according to the standard (i.e. recommendations), but in the end, they are not compatible with each other. Moreover, there is no one to complain to. In this regard, it is always necessary to check not the set of functions that is available on this or that VoIP gateway, but the set of standards supported by the gateway. In addition, the hardware manufacturers are saying that everything is compatible and SIP is unified – in practice, this is not the case, as each manufacturer strives to create its own unique service.
This leads to the fact that manufacturers create their own standards and these standards become bigger and the compatibility problem does not become easier. Because of this, when buying a VoIP gateway, it is better to check on the site of the manufacturer station, what gateways your IP-PBX supports and what restrictions are there.
Separately, we can mention the support for the H.323 protocol, which is still used. I will not go into the compatibility of H.323 in this article, because the protocol is not developed anymore, and no manufacturer develops it, except that it is only supported by the manufacturers.
There is also compatibility on the part of TDM. It would seem that TDM and analog telephony are standardized so that there should not be compatibility problems, but in practice, there are many questions, starting with pulse dialing, the correct definition of tones in the line and ending with different ISDN fields.
Term of support
VoIP gateway is an infrastructure device, and its feature is the physical limitation in performance. The data networks grow and more and more productive equipment appears, that supports 100 Mbps, 1000 Mbps, 1 Gbps and so on. In telephony, one line with one simultaneous connection is connected to one analog port, the E1 stream has 31-time slots and supports 30 or 31 simultaneous connections, depending on the type of signaling. This has not changed for many years and will never change.
On the other hand, there is VoIP, where everything changes quickly and dynamically – a new IP-PBX is emerging, new services and standards are emerging, unified communications and so on. When dealing with the network equipment, when new services appear, you usually do not have a question why you need more bandwidth of your IP channel and accordingly you change routers/switches. Then it is not clear why when changing IP-PBX or simply updating the IP-PBX you need to change the VoIP gateway. After all, you do not change the type of connection to the telephone network and the IP-PBX still supports SIP. Moreover, it turns out that the VoIP gateway manufacturer has removed your model from production and created a new model while the old one does not support him anymore. And sometimes the lifetime of the model can last 1-2 years, which for such a conservative world as telephony, is negligible. I agree that the equipment is becoming obsolete, production technologies are changing, but for us the support term of VoIP gateway is very important.
Installation requirements and type of cooling
This is more a requirement for analog gateways with FXS ports. One of the advantages of IP telephony is that the VoIP gateway can be placed as close as possible to the location of the phone. But the closer the equipment to the installation site, the higher the installation requirements:
- Passive cooling: probably, no one likes when the fan blows under his ear. However, this is not the only problem. You will not check the operation of the fan every day, while the support for remote fan monitoring on such models is not provided. And if the fan breaks, the gateway itself will break down quite quickly. Moreover, this will not be a guarantee case. It should be understood that a gateway with FXS ports requires electricity for each port, and from this, the cooling solution, with increasing ports, becomes only more difficult.
- Working environment: Naturally the gateway, in this case, is not in the server room and those temperatures that are in the room you can not control. It is also important to look more carefully at those operating temperatures that the manufacturer indicates.
Probably, each of us was faced with the problem that the quality of assembly or the quality of plastic of very cheap manufacturers leaves much to be desired. But if the build quality can lead simply to the fact that somewhere a screw or a breakage of ports is broken off when pulling out the cables (naturally random). That quality of plastic, which is so loved to save, can turn into the worst. As I wrote earlier, cooling is an important factor in the VoIP gateway, and if the quality of the plastic is good or a steel case is used, only the device will break if the device burns out. But if the quality of the plastic leaves much to be desired, then in addition to the fact that it can smell during work, what it can do while writing a fan, I think it is not worth it. And if you consider that the equipment is not in the server room, but somewhere near someone under the table, the results can be absolutely unpredictable.
Management and monitoring
Returning to the topic that the installation site of the equipment may be different, the requirements to remote management and monitoring are increasing. And if the manufacturers of small devices tend to support TR-069, to control the operator of communication, at the same time for installation in the infrastructure of the enterprise SNMP is always required and even better the possibility of a console cable.
This is probably the very important factor, but which everyone does not pay attention to when selecting, but only when operating. When testing the gateway, as a rule, it passes in “sterile” conditions. When the gateway is put on the network, it turns out that the quality is not as desired – there is an echo, unnecessary noise, faxes do not go (although the T38 is supported and the faxes are checked).
The reason is usually not in the wrong codec used or the telephone is changed, the reason lies in the gateway itself and its DSP processor. In fact, the reason is how good that DSP processor and its code can convert the RTP stream that comes from the IP side. Also, a very important factor is the functions of noise reduction, echo-compensation, signal level control, determination of pauses and other, purely voice, telephone functions. If this functionality was previously assigned entirely to the telephone exchange and all this was done in one place, now the VoIP terminals perform this all. And the quality of DSP will greatly depend on the quality of voice in your VoIP network. This option is more subjective, but in practice, users feel it very objectively.
Just like IP-PBX manufacturers, VoIP gateway manufacturers are trying to provide their users with additional services. We will only say that we advise you to pay attention to these services when choosing a gateway because they can be very interesting for your infrastructure and will give to you, something that no other producer will give. Who knows, maybe this will eventually become the most important plus for you.
From the above, we want to add that we should not consider as the main criterion – the price and efficiency of the basic call, but pay more attention to the little things. They make a quality product really high quality. And it turns out, like many operators, the cost of services falls slower than the quality of these services.