Different people periodically turn to me with the request to design telephony in offices, but then almost always the question comes down to integration with the data network and ultimately it turns out that half of the “telephony” project is eliminating errors and problems in the data network.
In fact, voice transmission is just one type of data, like a picture, or a letter - a digit everywhere. By and large, the global differences between the “telephone network” and the “Internet” are only in the beliefs of people and our inadequate outdated legislation ... well, and as a side effect of the stupid laws and greedy officials, are monopolists and “regulation”.
The same data is flying in the fiber, the only differences are in the protocols (ie, packet / frame headers). The networks are already very tightly integrated and disabling the packet transfer systems will instantly disable the telephone network.
General beautiful solution: a local network with Wi-Fi access points to which the telephone management system (tsiska or server) and the media gateway are connected to dock with the old telephony (a separate piece of hardware, or modules in tsiska).
So, how to integrate networks in medium office projects.
Let's start with the head, i.e. control systems. I worked with four systems: Cisco CME, Asterisk, MS OCS, frivolnyh SIP proxy (Siproxd, SER). Stable to such a level that it was not terrible to put in production, I saw only two: CME and SER, and the clients' demands and beliefs always uniquely define the system.
Cisco CME. There are two ways to deploy it: honest and expensive - buy a tsiska and a bunch of licenses, the minimum start is about $ 4k (new, official way), of which about $ 2k for iron. And the second way is to add one more sin to your soul, in this case only iron is bought. There is a completely “sinful” way: install Dynamips on one of the servers (better even with the entire GNS3 package) and load the required IOS into it, run “emulated tsiska” on it. In production, I never made a decision on the emulator, but in the lab and on the home compic to control one phone - it works fine, not worse than the iron one. I checked the work in the rented VPS in the e-StyleISP data center (http://www.e-styleisp.ru), connect the cell phone from home (via Corbin) - the voice quality is very good. On the same tsiska, the SIP Registrar is also launched, to which mobile phones are connected via a wi-fi access point, or the Internet. On a tsiska it turns out to connect fully phones and to achieve reliability of work such that for years not to remember about it. IVR services are also very easy to make (via vxml or tcl). Another advantage is that there is a lot of documentation and mid-level specialists; the deployment of complex services is real. Of the minuses - the difficulty of authorizing calls (it becomes difficult).
Particularly pleased with the use of software tsisko-background on Windows, the product is called "IP Communicator" on the laptop Samsung NC10. When choosing a laptop, I looked only with a Bluetooth stack from Broadcom - profiles for connecting headsets to a computer are best organized in it, in most other bluetooth stacks you can’t connect a regular headset at all - I threw out two USB adapters, just because they refused to see audio able stacks. Now Communicator is working on my beech in the background, it is connected in parallel to my work phone, it is very convenient to connect via VPN remotely. I talk through the headset, walking around the beech within a radius of 10 meters. But, like all tsiska, licenses for it cost money, about $ 150.
There is also a SCCP software client on Nokia, allowing you to connect to a tsiska as a native CP-7970, but it costs even more and also requires “activation”, i.e. not like other licenses for tsiska (just lying in the closet paper), but until you activate it - will not work.
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Docking with the telephone network is done correctly only through digital streams, i.e. E1 / ISDN, as an alternative - connect to the operator via SIP / h.323. ALL operators have the appropriate equipment, as a rule, laziness and procedural business hinders. If the connection is via the Internet, then at least on your side of the connection, configure QoS and ask the provider to check the counter QoS settings. The quality of VoIP over the Internet is determined solely by the quality of the work of technical providers of providers (everyone who goes through your traffic) and the leadership policy of “crush / not crush” someone else's VoIP. And in real life it boils down to only two indicators: the percentage of losses and jitter (variation in packet delivery time), works well with losses of less than 0.5% of packets and jitter less than 20 ms. The settings of the gateways can achieve stable operation with jitter up to 300 ms, but this will add a delay of 0.6 seconds to the response time (as in a mobile phone, the delay is not very comfortable). If the second subscriber on the mobile phone, or intercity - will be completely uncomfortable.
The next management system is Asterisk. I deployed in a virtual machine in the e-StyleISP data center, a distribution of
www.trixbox.org , but this is already in the following text.