The company that previously traded as "Feptias Limited" has now
changed its name to "Smartvox Limited".
The name was changed to more clearly reflect the business's focus in IP telephony. This web site
is unlikely to have the most up-to-date information about the company or its products and you are
advised to visit the new web site instead at:
The nearest equivalent to this page offering support and technical advice is here:
SIP is a highly versatile protocol and it can be used to establish a variety of
services that extend beyond VoIP to video, Presence and beyond. Even within the
voice environment, SIP can be used to implement various services that add functionality
and features on top of the basic end-to-end voice connection. This range of functions
is reflected in the surprisingly large number of so-called "SIP Server" types. In
this context the term "SIP Server" describes a particular service that can be implemented
using SIP rather than being a reference to a physical device (a PC for example).
One hardware device, such as a Dell or HP Server, can be used as the host for several
different SIP Server services. This should make more sense as you see what types
of SIP Server there are:
A SIP Proxy Server acts as an intermediate router for SIP requests and is always
located somewhere in the path between two end points. A SIP Proxy Server would never
be the end point itself. When it receives a SIP request it will examine it and decide
whether to pass it on unmodified, change it in some way and forward it, re-route
it to an alternative destination (or destinations - see "Forking Proxy below") or
reject it. When it rejects a request, it must send back a response giving a reason.
One of the core functions of a Proxy Server is to ensure that a SIP INVITE request
is routed to the correct onward destination. It would normally have a "dial plan"
for this purpose. The actual destination might be a PSTN gateway, another Proxy
Server, a SIP phone (often referred to as a User Agent) or a Media Server.
This is just a specialised type of Proxy Server that duplicates the SIP
request and forwards it to multiple locations. It must also handle the multiple
responses and filter them to ensure that the device that sent the original request
is not confused.
This is another specialised type of Proxy Server that responds to SIP INVITE
requests by returning a 3xx response. Alternatively, it may reject the call and
send a response to this effect. The main 3xx responses are 300, 301 and 302. Along
with with the 3xx response, the Redirect Server sends back one or more alternative
addresses using the SIP Contact header. The device that initiated the call and sent
the original INVITE receives the 3xx response and knows that it must re-try the
call using the address (or addresses) provided by the Redirect Server.
Registrar Server (and Location Server)
A server that accepts and handles SIP REGISTER requests. It is normally
combined with the function of Location Server - a Location Server simply being a
database which stores information about the current location of a SIP User Agent.
It is normal for a SIP phone - or other SIP User Agent - to register itself with
one or more Registrar Servers as soon as it is is switched on. The registration
will almost always involve authentication with a User ID and password. The User
ID for many SIP devices is the same as its phone number. Once registered, it is
easy for calls to be routed to the device provided the call passes through a Proxy
Server that has access to the Location Server. Without this registration process,
your IP phone would have to always be located in the same place using the same IP
address and that IP address would have to be hard coded into the dial plan of the
device calling it (or in the dial plan of a Proxy Server used by the calling device).
This is the name given to automated call processing servers (also known
as IVR's) in the VoIP World. A Media Server acts as the end point for a SIP call
and typically plays pre-recorded voice announcements and prompts, accepts DTMF key
presses or spoken instructions to allow it to implement interactive services such
as voicemail, tele-banking, tele-voting etc.
This term describes a service that acts as a transparent bridge, between two networks,
for media streams in a VoIP call. Servers that offer far-end NAT traversal often
use an integrated Media Proxy. They act as a Proxy Server, but modify the contents of the SIP packets (in both directions) to make it look as if the terminating device
requires the media stream to be opened at the address of the Media Proxy rather
than at its own address. This means the device that initiated the call and the one
that answered it have a media stream connection that goes
via the Media Proxy which is sitting between them (transparently proxying the media). The Media Proxy is always on a public IP address which means devices behind
NAT are able to initiate connections to it where they would not have been able to
initiate connections to each other.
Gateway Server or PSTN Gateway Server
If you want to make calls from your VoIP phone to a phone on the conventional telephone
network then your call has to be routed through a PSTN Gateway Server. The same
is true if you want to call a mobile phone, although it is possible for the mobile
service provider to make this gateway available as part of their infrastructure
rather than requiring all VoIP calls to first break out to the PSTN and then onward
to the mobile network. PSTN stands for Public Switched Telehone Network and it the
name given to the traditional telephone infrastructure that serves all the landlines
and telephones such as are still to be found in most homes. Most businesses still
rely almost entirely on the PSTN for their inbound call traffic and only use VoIP
for some outbound calls and for internal calls within the organisation (perhaps
between branches that are geographically remote but which have existing high speed
links for their data network).
B2BUA (Back to Back User Agent)
In some ways, this is like a cross between a Proxy Server and a Gateway
Server. Technically - in SIP terms - it acts as the end point for an incoming call.
In this respect it is unlike a Proxy Server. However, it would make use of a dial
plan to determine where to forward the call and would initiate a new call to the
required destination then links the audio paths together so that both remote parties
can talk to each other. You could say that a Gateway Server is therefore a B2BUA,
but strictly a B2BUA must be linking two VoIP calls together whereas a Gateway is
linking a VoIP call to a conventional PSTN call. To summarise, a B2BUA acts as the
"man-in-the-middle" for a SIP call path and appears to be the end point for both
call legs. This means that it does not forward SIP requests and responses like a
Proxy Server, but it does allow calls to be routed onward under control of a dial
plan and it links the audio streams of the calling and called parties so they will
think they are connected directly to each other. You may be wondering why this is
useful - the answer is that it can greatly assist in allowing calls to penetrate
networking boundaries such as between the Internet and a private LAN. It can also
be used to modify the encoding of the speech on each leg of the call if the B2BUA
is capable of so-called "transcoding". Asterisk is an example of a B2BUA capable
SBC (Session Border Controller)
This term is sometimes used to describe a server that incorporates some
of the services described above in such a way as to provide control of VoIP connections
at the edge of a private network where it meets the Internet. It's primary function
is to control and facilitate access to VoIP devices on the network that it protects.
However, some manufacturers also describe far-end NAT traversal devices as SBC's
which is probably a mis-use of the terminology, but then again the definition is
somewhat ambiguous. If it is acting as a far-end NAT traversal device then it would
either have to incorporate a Media Proxy Server or it would have to act as a B2BUA.
In fact, for an SBC to work as a B2BUA with two interfaces - one connected to the
Internet and the other to the protected subnet - makes a lot of sense. Firstly,
it overcomes problems of routing and IP address substitutions for SIP packets traversing
between two different networks, but it also permits a high level of control over
calls that are attempting to enter or leave the protected network. Asterisk can
be used very effectively in this role.