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How Does it Work?
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| The new way to communicate |
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Person-to-person communication is the new way of communication. Internet started as a small network accessible to a selected few. There are
two applications that have spread the Internet usage to so many people: email and web surfing.
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However, these applications do not direct real-time communication between individuals, a capability that is becoming highly useful as more and
more individuals have broadband or a fixed connection to the Internet. Now, real-time person-to-person communication is rapidly taking a position as one
of the most common uses of the Internet. This includes:
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Voice (of which IP telephony is but one component)
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Video communication
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Presence
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Instant messaging
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Conferencing with voice, video and data collaboration,
and much more
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| SIP - the standard that will bring us all together |
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Several forms of person-to-person communication over the Internet have already been in use for a few years. However, it is now, when a general
standard has been established, that these types of applications will become more available and more widely used. SIP is the Internet standard for such
applications and currently has a strong, accelerating growth.
A powerful driving force for SIP is that Microsoft has announced that all future real-time communication (RTC) will be based on the SIP standard. Windows
Messenger, which can be downloaded at no charge, already has a SIP mode that provides the user with telephony, voice, video, presence, and instant
messaging. And, Microsoft has recently launched the Windows Real Time Communications Server 2003. It includes a SIP server for enterprise usage and a
programming API, which is expected to result in numerous SIP applications. With the market impact of Microsoft, the already large number of SIP users will
grow even further.
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| IP Telephony – just one part |
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SIP can also be used for "ordinary telephony", i.e. voice with 3 kHz bandwidth and common number dialing, over IP networks. For this application, the SIP
standard is taking over from H.323, which is a protocol from the standardisation organisation of the telecom world, the ITU-T. H.323 has been used
to build islands of VoIP, but most often without interoperability on the IP level between the different operators. Another protocol is MGCP, or the related
H.248/MEGACO, that sometimes is used to control IP phones on a low level in order for operators to connect these to the old telephone network, the PSTN.
It should be noted that IP telephony - where ordinary telephony is emulated over IP - is only a small part of person-to-person communication for which
SIP was created. Real-time person-to-person communication involves so much more than just telephony as mentioned earlier.
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| Why hasn't the old telephony network been replaced by VoIP? |
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Traditional firewalls are the main obsticle for Internet based real-time communication. Most firewalls installed today do not handle SIP in an adequate
way. The problem occurs when a person on a private LAN is to be contacted. Ordinary firewalls are simply not designed for such data traffic.
It is a common misunderstanding that well known firewalls can be configured to handle SIP traffic, but that is not the case. One problem is that the media
streams (e.g. voice and video packets) are transferred over dynamically assigned UDP ports that are generally closed. Another problem is that the SIP
clients inside the firewall cannot be reached by IP addresses since these most often are private and local to the LAN. It simply does not work, unless there
is specific SIP support in the firewall.
The same applies to routers that are changing the address space, NATs. NAT routers are used when several users share a common Internet connection with a
single IP address.
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Enabling NATs and firewalls with SIP
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That person-to-person communication does not reach the users on the LANs is of course a fundamental problem. Various methods and equipment
have been suggested to solve this problem in a number of situations, but the most general one is to eliminate the problem where it occurs - in the firewall
itself. Firewalls including a SIP server (with a built-in SIP proxy and SIP registrar) that dynamically controls the firewall are currently available.
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Intertex foresaw the future impact of SIP and the firewall/NAT problems as early as in 1998 and started developing the current architecture of the
Intertex Internet Gate line with this in mind. This led to the creation of a firewall and NAT with a built-in SIP proxy and SIP registrar, the most general
architecture for SIP handling.
The SIP proxy dynamically controls the firewall, opening and closing ports as needed. The SIP registrar keeps track of all the users inside the firewall
and makes sure that the communication is directed to the correct device. Even more remarkable is that the SIP registrar in the Internet
Gate can be used as the main SIP server for a company and can even be configured to handle registrations from the outside, making it ideal for road warriors
and telecommuters.
Furthermore, with the built-in proxy and registrar it is possible to handle encrypted SIP signaling, a feature not accomplished by
other architectures.
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SIP capable firewalls are not more expensive than ordinary firewalls and should be considered for all new installations of
firewalls and NATs. If not, there is a high risk that even newly installed firewalls and NATs have to be exchanged in the near future.
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| The SIP Switch |
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By upgrading the Intertex Internet Gate with the SIP Switch you will get a number of PBX-like features and it will conveniently integrate the combined
usage of soft SIP clients, SIP telephones and ordinary telephones. Important features include local extension numbers, dial 9 (or 0) for "outside line",
forwarding, forking and other advanced features.
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| ENUM lookup is also an important part of the SIP Switch. This is used to check whether a phone number has a SIP address, before handing a call over to a
PSTN gateway. All these features are essential for the coming migration from PSTN to SIP-based communication for small offices and home offices. |
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If you have an IP telephony account from a SIP service provider this can usually be integrated with the SIP Switch, allowing all users of the SIP Switch
to benefit from the low rates offered for calls to ordinary telephones. In addition, by integrating accounts from multiple countries, you can create virtual
foreign offices, allowing customers and partners to call the SIP Switch users on local telephone numbers only paying local rates. This will also drastically
reduce the cost for outgoing international calls to the countries where you have local accounts.
The SIP Switch combines the best of two worlds; URL dialing, presence, instant messaging and video from the IP world can be used simultaneously with number
dialing and traditional PBX functions required to make SIP hardware phones convenient to use. The SIP telephones work just like ordinary telephones with full
access to and from the PSTN, using ordinary phone numbers without any limitation of the IP SIP functionality. |
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