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Registering with a SIP Server or PBX
The process of registering a device to a SIP provider, whether it’s in the “cloud” or at your location, is usually simple. Much
like registering an email client with a mail server, the VoIP client (the VoIP hardware) must know the location of the server,
and a username/password combo with which to register. The server location can be in the form of an IP address, or a URL.
Some servers with more complex arrangements may require more information to help choose options. There may be separate
settings for your SIP Proxy server, your SIP domain, and your SIP registration server. There may be choices for encoder support,
auth username (an additional credential used for authentication), and caller ID options. For the most part, any essential info
that needs to be programmed will be delivered from your provider (or in the case of a PBX, your Telco department) and you
can set your VoIP device with the parameters that match, and ignore the others.
Making and Receiving calls
Once registered correctly with a SIP server, incoming calls will be routed to your SIP device based on the calling plan set up
with your provider or PBX. Whether it’s the DID line(s) assigned to you by the provider, or an incoming trunk attached to your
PBX, a “ring” on the line will trigger the server to notify your device of a call request using the SIP protocol. Your device can
accept or reject the call. If you accept the call, an RTP channel is created to your device each way.
Outgoing calls just reverse the process. The SIP device sends an outgoing call request to the server, which attempts to
complete the call. Call progress messages will be sent to your SIP device from the server, which may translate them to familiar
tones like ringing and busy. On call completion, the server will create the RTP channels in the same way as for incoming calls.
Hunting
Of particular interest to broadcasters who take lots of calls simultaneously is hunting behavior, or the way the system behaves
toward simultaneous incoming calls. Keep in mind, when an incoming call is in the “ringing” state, there are only status
messages exchanged over the SIP connection--no actual audio is being transferred. The RTP audio channels are only created
after the call is answered.
Only one SIP connection needs be open for multiple voice channels to be created. Your VoIP provider or PBX will be
programmed to allow a designated number of simultaneous voice channels, and any further incoming calls will be rejected
there. By default, most multi-channel VoIP gear will “hunt” any second, third etc. call to the next “line” on the device. In this
way hunting is inherent. If more than the supported number of calls is requested to the VoIP device, it will reject them in the
same way as the provider does, and no RTP channel will open for these excess calls.
Alternately, it’s possible to set up a separate SIP account for each “line” on the SIP device, and this account should be
capable of creating only one “channel” at a time. In this case, it’s the responsibility of the provider or PBX to sort the hunting
arrangement and notify the proper account about incoming calls.
Choke Lines
Another topic of interest to broadcasters is choke lines, the specially conditioned telephone trunks designed not to fail under
loads of thousands of incoming calls (e.g. for contests). In the PBX scenario, choke lines can easily be used as the trunks that
feed the PBX, and very little changes.
When using a cloud provider, it’s important to notify them about potential peak call volume to avoid overloading their systems.
But cloud providers are usually equipped to provide service to high-volume nationwide call centers, so they can usually
implement techniques to throttle large amounts of calls without impacting overall service.
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