What SIP server/registrar should I use?

Discussion in 'VOIP' started by Ramon F Herrera, Jul 31, 2005.

  1. I am assembling my own SIP VoIP infrastructure, and have the endpoints
    ready: a fancy Cisco and digital PRI at the office and a Sipura 2100 at
    home. The only remaining part of the puzzle is the SIP registration
    server. I am trying to determine what servers are available and which
    is the most convenient. A quick research turned up these potential
    candidates:

    * Asterisk
    * SIP Express Router: An Open Source SIP proxy/router
    * OpenSER: GPL SIP server

    Question Number 1: Can Asterisk be a SIP registrar for non-Asterisk
    calls?
    Should I use a dedicated server instead?

    TIA,

    -Ramon F Herrera
     
    Ramon F Herrera, Jul 31, 2005
    #1
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  2. Ramon F Herrera wrote:
    > I am assembling my own SIP VoIP infrastructure, and have the endpoints
    > ready: a fancy Cisco and digital PRI at the office and a Sipura 2100 at
    > home. The only remaining part of the puzzle is the SIP registration
    > server. I am trying to determine what servers are available and which
    > is the most convenient. A quick research turned up these potential
    > candidates:
    >
    > * Asterisk
    > * SIP Express Router: An Open Source SIP proxy/router
    > * OpenSER: GPL SIP server
    >
    > Question Number 1: Can Asterisk be a SIP registrar for non-Asterisk
    > calls?


    Yes. And it can do a lot more: it can register on other servers, acting
    as client; route calls, handle voicemail, conferencing etc. like a PBX;
    translate between different protocols (SIP+RTP <-> IAX) and codecs
    (G.711, GSM, iLBC etc.)...

    > Should I use a dedicated server instead?


    It depends on the amount of traffic you want to handle: some large
    sites such as FWD use a combination of SER and Asterisk. But if it is a
    small infrastructure, even a single registrar might represent overkill:
    you could e.g. use the free service of Like2Fone (www.like2fone.com )
    or FWD. However, this would assume that you may work around NAT issues,
    which can represent a thorny issue depending on your NAT router. Hint:
    if you have more than one phone on the same NATted LAN, make them
    listen on different UDP ports, and on each enable both STUN and
    symmetric RTP.

    Alternatively, if you can, you should run Asterisk on top of the NAT
    router, which is possible if the latter is a Linux machine or a Linksys
    WRT54GS reflashed with OpenWRT. If Asterisk binds to the address
    0.0.0.0, it will listen to both internal and external interfaces
    without any natting: so the phones on the LAN will register on it, and
    in turn it will register on any number of external registrars.

    It is also possible to run Asterisk behind a NAT, but as Asterisk
    doesn't support STUN, you'll need to configure its SIP service telling
    it the external IP address of your router and the local network
    IP/netmask. This may be problematic if your Internet connection has
    dynamic IP address! In that case, you may be avoid a lot of headaches
    giving up SIP for the connections between Asterisk and the rest of the
    world, and use IAX. This will still allow you to connect to a number of
    free providers (e.g. FWD) and commercial PSTN termination providers
    (Voipjet, Voxee and others). Your phones may still use SIP to talk with
    their local Asterisk(s).

    Enzo
     
    Enzo Michelangeli, Jul 31, 2005
    #2
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