What you need to know about SIP – Session Initiation Protocol

What is the right notation – IP Phones (or) SIP Phones? SIP, has become such a common protocol for voice communications over IP that we tend to think that SIP phones are IP Phones! What is so special about SIP that enabled it to beat the H.323 implementations of the biggest telecommunication companies? Let us find out, in this article.

What is SIP?

SIP, refers to Session Initiation Protocol. SIP is an IETF open standard scalable and extensible protocol for carrying voice, video and multimedia communications over the IP Network.

The Session Initiation Protocol simply provides interoperability between the various components of the telecommunications equipments as it is an open standard (vendor neutral) protocol. Its because of SIP, that today you are able to go out to the market, buy an IP Phone and expect it to work in your home or company! Prior to SIP, it was proprietary implementations of H.323 that ruled the market. It means, each big telecommunications vendor had their own version of the VOIP protocol (based on H.323) and only the IP Phones and the PBX (VOIP Switch) that confirmed to that protocol (read: same vendor) could inter-operate with each other. There was nothing technically wrong with this approach, just that people were used to buying a variety of phones from the market for using with their TDM based EPABX and didn’t like the vendor-locking scenario for IP Phones!

Phones are just one small part of IP Telephony systems. To understand the real power of this protocol have a look at,

Some of the applications supported by SIP:

• Most of the features of a PBX (Call Ringing, Call Pickup, Call transfer, auto attendant (IVR), voice mail, etc are supported by SIP
• Voice Mail to Email and vice versa
• Click to dial from web pages
• Meet-Me conferencing with a large number of users/ rooms
• Unified Messaging – Integrating voice, fax and email in a single mail box.
• Presence (Availability) Status broadcasting and receiving – like Instant Messenger (Available, Busy, etc)
• Presence based Call Routing, One Number Dial
• Integration with AAA Servers like Radius and Industry standard directories like LDAP
• Text to Voice and Voice to Text
• SIP is supported by a variety of end points like Desk Phones, Soft Phones, PDA’s, Cell Phones, Wi-Fi Phones etc
• Business Process Integration (Time of the day routing, CRM Software integration etc)
• Integration with Web Applications (Like Google Maps)
• Video Conferencing, and a lot more….

So, basically SIP supports a modular philosophy, itself focusing on a specific set of functions only. For achieving that modularity, SIP reuses and integrates with a number of existing protocols like SMTP, LDAP, IMAP, SOAP (XML), HTTPS, XML, POP etc. SIP was developed by the IETF (Internet Engineering Task Force) which developed other popular web based protocols like SMTP, HTTP, etc and as it is web based, it offers faster development cycles, vibrant community of developers and integration with web based applications.

To further illustrate its similarity with HTTP and SMTP protocols, let us look at a SIP address (URI) – sip: name@company.com. As you can see, this is similar to a website address or an email address. Further, all SIP messages are text based messages (like INVITE, BYE etc) that are readable.

SIP operates in the application layer of the network – it can establish, modify and terminate multi-media sessions between intelligent devices and hence it is not restricted to voice. It can be used for video conferencing and even IP multimedia streaming in cellular systems.

SIP uses SDP (Session Description Protocol) for signaling and RTP (Real Time Transport Protocol) for media transport. SIP can be classified as a Peer to Peer protocol as two intelligent devices based on SIP can communicate with each other by themselves (SIP Phones, for example) and even when a call control unit is used (IP PBX), SIP coordinates for the initial call set up (using SDP) and later on, media traffic flows directly between the various end points (using RTP). So, more control is moved on to the end points through SIP, which makes the endpoints ‘intelligent’ and hence perhaps, the higher cost!

To have a look at the various elements and architectural components of SIP, click here.

SIP Trunking:

1. There are two types of SIP Trunks – SIP Trunks between two or more Enterprise IP PBX (Private Switch)     &
2. SIP Trunks between Enterprise IP PBX and Service Provider Network (Public Switch).

As you would remember, the word Trunk is from a Telephony service provider background. Basically, if you need to call to the outside world, you need to buy Analog/Digital Trunks (Lines) from a Telephony service provider so that you could call any land line or cell phones across the world. You would most probably be charged based on the distance – Local Call, STD Calls and ISD Calls being the norm.

But look at the same thing in the IP Scenario. You still need to take a line (not a physical line, but a virtual one through Internet) form an ITSP (Internet Telephony Service Provider) and connect it to the IP PBX in your organization. For connecting this, you need a feature called SIP Trunking to be supported by both your IP PBX as well as the Service Provider’s Public Switch. It sort of integrates both the telephony systems (partially) and lets you use this IP Trunk to make calls to land line or cell phones around the world (where ever the service is supported by ITSP). But the huge advantage in this method is that, you pay reduced charges for your long distance calls (STD, ISD) as the calls are now carried over the IP Networks for the maximum distance by the ITSP. But do note that in some countries, this type of SIP Trunks are not allowed (or) restricted.

Now, take a scenario where you have two branches – One in Frankfurt and another in New Delhi. You have your corporate PBX set up in Frankfurt and another PBX in New Delhi. If you want to assign extensions like 301, 302, etc in Frankfurt (and) 401, 402, etc in New Delhi and want any person in either place to reach the others just by dialing their extension number, then both the PBX (IP or TDM) should be trunked to each other over IP WAN Networks (Leased Lines, MPLS, VPN Over Internet etc). So, SIP Trunks help you achieve just this and they can Trunk all the PBX’s of various branches of a company, as long as there is a proper WAN network available between them and all of them support SIP Trunking. This can also be done by using IP/ H.323 Trunking if all the PBX are from the same vendor but SIP Trunking can be done between multiple vendor PBX systems!

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