telephone systems that I crossed with VoIP Wi-Fi mobile phones. Cisco CME

    Different people periodically turn to me with a request to design telephony in offices, but then the question almost always comes down to integration with a data network, and in the end it turns out that half of the project “by telephony” is the elimination of errors and problems in the data network.
    In fact, voice transmission is just one type of data, like a picture or a letter - everywhere a long time ago figure. 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differences between the “telephone network” and the “Internet” are only in the beliefs of people and our inadequate outdated legislation ...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 stupid laws and greedy officials - monopolists and “regulation”.
    The same data flies in the fiber, the differences are only in the protocols (i.e. packet / frame headers). The networks are already very tightly integrated and disconnecting packet systems will lead to an instant disconnection of the telephone network.
    A general beautiful solution: a local area network with Wi-Fi access points to which the telephone management system (tsiska, or server) and a media gateway are connected to connect to the old telephony (separate piece of hardware, or modules in tsiska).

    So, how to integrate networks in mid-office projects.
    Let's start with the head, i.e. management systems. I worked with four systems: Cisco CME, Asterisk, MS OCS, frivolous SIP proxies (Siproxd, SER). Stable to such a level that it was not scary to put in production, I saw only two: CME and SER, and the requirements and beliefs of customers always uniquely determine the system.

    Cisco CMEThere are two ways to deploy it: honest and expensive - buy a tsiska and a bunch of licenses, the minimum start is about $ 4k (new, by official means), of which about $ 2k for iron. And the second way - adding another sin to your soul, in this case only iron is bought. There is a completely “sinful” way: install Dynamips on one of the servers (it’s better even with the entire GNS3 package) and load the necessary IOS into it, run the “emulated tsiska” on it. In production, I never made a decision on the emulator, but in the lab and on the home computer to control one phone, it works fine, no worse than iron. I checked the work in a rented VPS in the e-StyleISP data center (http://www.e-styleisp.ru), I connect the mobile phone from home (through the corbine) - 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 tsiska, it turns out to connect the phones fully and achieve reliability so that for years not to remember about it. IVR services are also very easy to do (via vxml or tcl). Another plus is a lot of documentation and mid-level specialists, the deployment of complex services is real. Of the minuses - the difficulty of authorizing calls (difficult).
    The application of software cisco-background on Windows was especially pleased, the product is called "IP Communicator" on Samsung NC10 laptop. When I selected a laptop, I looked only with the Broadcom Bluetooth stack - it has the best organized profiles for connecting headsets to a computer, in most of the other bluetooth stacks you can’t connect a regular headset at all - I threw out two USB adapters, just because they refused to see the audio able stacks. Now I have Communicator running on a beech in the background, it connects in parallel to my work phone, it is very convenient to connect via VPN in the remote. I talk through the headset, walking around the beech in a radius of 10 meters. But, like all tsiska, licenses for it cost money, about $ 150.
    There is also a SCCP software client to Nokia, allowing you to connect to tsiska, like the native CP-7970, but it costs even more and also requires “activation”, ie not like other licenses for tsiska (just pieces of paper lying in the closet), but until you activate it, it won’t work.

    Docking with the telephone network is done correctly only through digital streams, i.e. E1 / ISDN, as an alternative - to connect to the operator via SIP / h.323. ALL operators have the appropriate equipment, as a rule, laziness and procedurality of the business interfere. If the connection is via the Internet, then at least on your connection side, configure QoS and ask the provider to check the oncoming QoS settings. The quality of VoIP over the Internet is determined solely by the quality of the work of the technical specialists of the providers (everyone through whom your traffic goes) and the policy of the “push / no push” guide for someone else's VoIP. And in real life it comes down to only two indicators: the percentage of losses and jitter (variation of packet delivery time), works qualitatively with losses of less than 0.5% of packets and jitter less than 20 ms. With the gateway settings, you can achieve stable operation with jitter up to 300 ms, but this will add a delay of 0.6 seconds for the response time (as in the mobile phone, the delay is not very comfortable). If the second subscriber is on a mobile phone, or intercity - it will be completely uncomfortable.

    The next control system is Asterisk. Deployed in a virtual machine in the e-StyleISP data center, the distribution www.trixbox.org , but this is in the next text.

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