Migration to VOIP System

Posted on 2007-10-02
Last Modified: 2013-12-14
We have 3 T1 lines

two for internet,
One for PRI Voice

i am looking into VOIP but still a beginner. What are the options for VOIP migration? With that could i eliminate the extra voice line? i was looking at previous posts at ee, seems like Cisco / Avavya products are at enterprise level - Also came across SKYPE - but skype seems to be more home based than enterprise - correct me if i am wrong.

Question by:rdvarghese
    LVL 36

    Expert Comment

    Skype is purely home based and uses p2p technology and the disadvantage of this is that it is practically impossible to implement any quality of service which makes it unsuitable for corporate use.

    With VoIP you could eliminate the extra voice line but it depends on how reliable your internet connection is. You need a perfect internet connection between yourselves and the VoIP provider.
    Generally I recommend keeping the PRI line for incoming calls and general outgoing calls. Then you can use the internet connection with VoIP for any calls between offices or any locations which are much cheaper over VoIP.

    Cisco and Avavya are good. 3COM do some good equipment also.
    I use Asterisk which is an open source PBX running on Linux. It is oviously much cheaper but you do need experience in administering Linux systems and configuring Asterisk.
    LVL 11

    Accepted Solution

    I do not recommend to drop PRI. Public Internet is not reliable media, so even if you feel like it's ok, for voice traffic it might be not acceptable. Just look at the requirements for network to implement enterprise grade VoIP services.

    One way delay <150ms (250ms still acceptable)
    Jitter (variation of delay) <30ms (actually 10ms)
    Loss <0.5% (single packets only)

    Voice packets create consistant traffic which doesn't tolerate temporary interruptions. VoIP uses UDP/RTP for voice transmission which doesn't recover any lost or delayed packets because there is no time for it. Packets arriving not in time will be just dropped by codec because there is no use for it, so if you will see that all the packets from point A made it's way to point Z it doesn't mean that you will hear crystal clear voice.

    In most cases if your ISP is VoIP carrier as well, then you can hope to make some arrangments with them such as priority queuing for voice traffic across their network.

    We use MITEL ICP 3300 which is much more flexible then Cisco, support is much faster and easyer to deal with and it's cheaper per seat if you use embedded voicemail and autoattendant. MITEL ICP does not reuire any additional components to run. They have different boxes for different number of users, so you don't need to spend all your money to buy a mega-system for 30 users at remote location and if youneed it can serve thousends of people with bigger boxes. It does all what Cisco and Avaya do but a little bit better and friendlier.
    Let's say, what will do Cisco phone if it looses contact with a server, right immediately restart regardless of existing calls. In MITEL world it will allow to end existing call if voice path wasn't affected. In resilient pair of controllers all the phones will switch between primary and secondary controllers without any interruptions, just a little indication on the screen. Besides that MITEL can be intergated with SIP systems, ISDN PRI/BRI, all sorts of analog trunks and analog phones and all this out of the box.

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