[Gllug] VoIP at home, softphone, interoperability

[Andy Farnsworth]

Peter Childs wrote:
>
>
> Some of you seam to think SIP is difficult the other half seam to 
> think its really really easy. This is a reoccurring subject on gllug 
> as well! I think a meet on VOIP (and Linux) might be a good plan.
>
> For those thinking SIP is complicated from what I've heard of TAPI its 
> worse. Try 33 operations just to create a new user and I got that 
> quote from a telephone sales guy.
>
> Peter.
My experience with SIP is two-fold.  First using an ATA (Analog 
Telephony Adapter) to connect a normal POTS (Plain old telephone 
service) telephone to Broadvoice ( a VoIP provider).  The second, 
connecting multiple Hard and Soft phones to a VoIP Server (Asterisk on 
Linux) with a trunk connection to Broadvoice.  My experiences varies 
slightly between the two:

1) ATA -> Broadvoice : brain dead easy
     Receive Preconfigured ATA from broadvoice, plug it in and it worked.
     2 years later the ATA Died so I bought a new one.
     Contacted Broadvoice support and they helped me configure it to 
point to their tftp server and I was up and running in 5 minutes.

2) Hard and Soft VoIP phones -> Asterisk -> Broadvoice
    Hard and Soft VoIP SIP phones to asterisk ON THE SAME NETWORK : 
Brain dead easy
        Point phones to the internal IP (or name provided DNS works) of 
the Asterisk server.
        Register new extensions in Asterisk to support the new phones.
        Give phones the appropriate username and password for each one.
        Phones register with Asterisk, no problem.

    Hard and Soft phones to Asterisk Outside my network : Much more 
difficult
        SIP does not do well with NAT, at least on the server side.
        Client phones work fine from behind NAT firewalls as long as the 
server is directly reachable via the internet.
        Asterisk Server connected directly to internet and phones on the 
internal network pointed at the externally exposed IP of the Asterisk server
        Phones register with Asterisk no problem.

     Asterisk -> Broadvoice Trunk : Not easy, but not too hard
        Enter username and password for Broadvoice account into 
templates, save these modified templates as active configuration files 
and restart asterisk.
        Configure Asterisk to monitor the trunk for incoming calls and 
route them to the appropriate extension.
        Configure Asterisk to allow outgoing calls on the trunk by 
prefixing any number dialed with 8,



Summary:
   Connecting a SIP Client (Soft or Hard phone) should be very simple, 
provide server IP / Name, Username, and Password and that should be al 
that is required.  Note that skype will require the exact same thing, 
except it already has the server name built into the client so it just 
needs username and password.

   Correctly configuring a SIP Server for what you want it to do is a 
more daunting process, but it is much more flexible and can provide you 
with a much more useful tool.  Some of which are listed here:
         Answer your incoming calls from POTS Landlines, VoIP Providers, 
registered extensions (hard or soft VoIP phones), other VoIP peers, etc
         Route these calls appropriately:
                Ask the caller to dial the extension of the party they 
are trying to reach (includes directory if you want it, Music on Hold, 
voicemail, etc)
                Hunt for the person dialed, for example, ring my desk 
phone first, if I don't answer, ring all my hard and soft phones that 
are registered to ME, if I still don't answer ask the person to leave a 
message or enter a security code and Asterisk will use my POTS line and 
dial my mobile and then link the caller to me that way.

                Ring all the phones regiersted to me, ring all phones 
registered, then send the caller to my voice mail box.

                Allow the user to dial back out of the system using one 
of the free trunks, useful if I am on my mobile and want to call to the 
UK using my VoIP trunk. 

         And many many more uses...

Back onto the specific topic, for what you seem to want to use it for, 
SIP may work but will probably cause you headaches because one or both 
of your devices will be behind a NAT firewall and you won't be running a 
server.    If you are running asterisk, then you can do anything you 
want and/or can dream of but you do have to figure out how to install 
and configure Asterisk or other VoIP Server.


Andy Farnsworth
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