[VoIP] SIP and Skype

http://www.tutorial-reports.com/internet/telephony/voip/sip/voip-skype.php 

0 – INTRODUCTION

Voice over Internet Protocol (VoIP), the revolutionary communication technology is all set to replace the the traditional phone system in the near future.

Today, it is possible to make toll free long distance voice and fax calls over existing IP data networks. Businesses implementing own VoIP systems can save a lot on long distance calls.

VoIP converts the voice signal from telephone into digital signal that travels over the internet and then it is converted back into voice signal at the other end, thus making it possible to speak using regular phones with Internet.

At each end of a VoIP call, there is an analog soft or IP phone acting as a user interface, client software working with a codec to handle the digital to analog conversion and soft switches mapping the calls. The different pieces of hardware and software work together with various protocols.

The protocol H.323 developed by International Telecommunication Union (ITU) was widely used earlier. It was however found to be very complex and consisted of many protocols and specifications. It was not specifically tailored for VoIP.

Then came Session Initiation Protocol (SIP), specifically for VoIP application. Media Gateway Control Protocol (MGCP) is the third commonly used VoIP protocol. Now there is a proprietary peer-to-peer protocol used in Skype.

Read on to find out how SIP and Skype are driving the future of VoIP.

1 – WHAT IS SIP

Internet and mobile communication have taken over the individual as well as the business communication methods on a large scale. The use of instant messaging applications is increasing year after year. In this scenario Session Initiation Protocol (SIP) plays a very crucial role.

A close monitoring of the telecom world will show that it is slowly shifting from separate voice and data networks to a single converged network for all forms of communication. All multimedia applications are usable over a common IP-based hardware/software infrastructure including both fixed and mobile elements and broadband transmission. SIP will equip the platforms for this IP Multimedia Subsystem (IMS).

SIP is an application layer protocol, RFC standard (RFC 3261). It was developed and designed within the IETF (Internet Engineering Task Force) MMUSIC (Multiparty Multimedia Session Control) as an alternative to H.323. SIP is a request-response signaling protocol (system of formal rules) for setting up and starting voice, video, and instant messaging communication sessions over Internet.

SIP, however, is independent of the details of the session. Sessions here mean a set of senders and receivers that communicate and the state they are kept in during the communication. SIP isn’t all in one; it just makes the communication possible – the communication itself is achieved by other means.

SIP is neither a session description protocol (SDP), nor does it provide conference control. The main advantage is that it is compatible with different architectures and deployment scenarios in the Internet services. Its essential communication function is aided by extensions and further protocols and standards. Two protocols commonly used are: RTP and SDP.

RTP (Real-time Transport Protocol) is used to carry the real time multimedia data like audio, video, and text. This protocol encodes and splits the data into packets and transports such packets over the Internet. SDP describes and encodes capabilities of session participants. Such a description is then used to negotiate the characteristics of the session so that all devices can participate.

SIP’s main function is to help session originators deliver invitations to potential session participants irrespective of where they are. To achieve this SIP uses a wide variety of protocols. Each protocol distinctly addresses the different aspects of the requirement. Some of the protocols are: SOAP, HTTP, XML, VXML, WSDL, and SDP. SIP controls the functioning of MGCP and MEGACO also. Some specific functions are:

a) Name translation and user location – ensuring that the called party is located and the call reaches it.

b) Feature negotiation – coming to an agreement on the features to be supported by the group involved in the call.

c) Call participant management – the user can bring other users in to the call or put them on hold or cancel their call while the session is on.

d) Call feature changes – enabling the user to change call characteristics like voice-only facility for one user, video function for another during the same session.

e) SIP also enables other capabilities like encryption and security.

1-1 – SIP PROTOCOL ELEMENTS

SIP has the following entities, each with different function.

1. SIP Terminal - This supports the real time, two-way communication with other SIP entities.

2. SIP User Agent - The user agents are the endpoints of the call. So there is the User Agent Client (UAC) initiating the call and sending SIP requests and then the User Agent Server (UAS) answering the call. It receives and responds to SIP requests and can accept, refuse or redirect the call. The User Agent software switches between the UAC and UAS modes on a message-by-message basis depending on what is going on. The User Agents can be handsets or desktop applications.

3. SIP Network Server - This network device handles the signaling associated with multiple calls and allows peer-to-peer calls to be made using client-server protocol. The main function is to provide name resolution and user location, and to pass on messages to other servers using next-hop routing protocols.

There is more than one type of server: the Proxy Server , Redirect Server , and the Registrar Server .

Proxy Servers are network hosts acting as both clients and servers to other entities. The job is to ensure requests are routed to appropriate entity identified by a SIP Uniform Resource Identifier (URI). The Proxy servers can operate in two different modes: The SIP stateful Proxy server and the SIP stateless Proxy server .

A stateful server remembers the incoming requests it receives and the responses it sends back and the outgoing requests it sends on. These stateful mode servers are the local devices close to user agents controlling domains of users.

A stateless server forgets all the information once it has sent a request. These stateless servers are the crucial part of SIP infrastructure.

Redirect Servers receive SIP requests and send response to zero or more addresses. The first location to answer takes the call. Redirect servers do not initiate SIP requests or accept SIP calls.

Registrar Servers accept registration requests. These servers maintain the databases that contain location information of all user agents registered with a particular SIP domain, thereby enabling the users to update their location and policy information.

4. Back-to-Back User Agents - A B2BUA is a combination of UAC and UAS and plays the role of making the response generated by the UAS part (to an incoming request), dependent on the response received by the associated UAC part (to a further request that UAC generates). Thus a B2BUA maintains information on the state of a dialog and participates in all requests sent on the dialogs it has established.

1-2 – How SIP enables VoIP

SIP enables the VoIP (Voice over IP) industry to produce inter-operable telephony products. Customers can now use an IP PBX from one vendor, a media gateway from another, and a phone from a third vendor. SIP is fast replacing H.323 in VoIP.

SIP uses session description protocol (SDP) as a media description language and RTP / RTCP as a real time transport protocol for media. SIP User Agents (UA) take the role of end points; the Proxy, Registrar and Redirect servers take the roles of network management, addressing resolution, authentication, and authorization.

How does this work?

The primary role of SIP is to allow the caller and the called party to contact each other, to set up, modify, and finally end various types of communication sessions like voice calls and video conferencing. The protocol components mentioned above together deliver messages embedded with the SDP protocol, defining their content and characteristics, to complete a SIP session.

The accompanying figure shows very simply how some of these SIP logical entities use messages to interact – in this case to set up a voice call from a PC (soft phone) to (hardware) SIP VOIP phone

 

Figure 1. SIP Session in dissimilar Domains (http://www.lightreading.com/)

The message and action sequence for establishing a SIP session in dissimilar domains is explained below. The following numbered points explain the red and blue arrows in the figure.

1. User X in Domain A calls User Y in Domain B
2. Query of Proxy server in Domain A: ‘How to get to User Y in Domain B’
   (Here the SIP Proxy Server of Domain A recognizes that the User Y is outside the Domain A and queries the SIP Redirect Server for User Y’s IP address.)
3. Redirect Server’s response: ‘Address is enclosed in the response message, send the requests to Proxy Server in Domain B’.
4. Call Proxied to SIP Proxy Server of Domain B.(SIP Proxy Server A forwards the SIP session invitation to SIP Proxy Server B)
5. Proxy Server B’s query: ‘Where is User Y?’
6. Registrar Server B’s response: ‘User Y is at the address enclosed in this response message.’
7. Proxy server B delivers User X’s invitation to User Y.
8. User Y’s response. User Y responds to User X’s call.
9. Response. Proxy server of User Y sends the response of User Y to Proxy server of User X.
10. Response. Proxy server of User X conveys User Y’s response to User X. (User Y forwards his or her acceptance along the same path the invitation travelled.)

If the call set-up is successful (Y is free to take the call), a media path using RPT is established between X and Y and the connected parties can start to talk.

2 – VoIP through Skype

For those of us who love talking, there is Skype. And when it comes talking for free, it is not a surprise then, that more and more people grab such facilities as provided by Skype.

Skype was founded in 2002, to develop first peer-to-peer (P2P) telephony network. Today Skype is competing against established, open VoIP protocols like SIP and H.323. VoIP with Skype software allows telephone conversation through a PC and over the Internet instead of a separate phone connection. This proprietary freeware , uses messenger-like client, and offers in and outbound PSTN (Public Switched Telephone Network) facilities.

Skype users can call to any non-computer based landline or mobile telephone in the world. Skype users can call other Skype users for free. The calls made to or received from traditional telephones are charged a fee and so are the voice-mail messages.

Skype software also provides features like voicemail, instant messaging, call forwarding, and conference calling. Skype users are not billed according to the distance between the two countries. Instead the users are charged according to the prosperity of the country, the volume of calls made to and from the country, and the access charges. Latest statistical figures show that Skype is one of the fastest growing companies on the Internet.

Skype Statistics

• Skype has 54 million members in 225 countries and territories and the number is swelling, all this by just word-of-mouth marketing by satisfied users!
• Skype is adding approximately 150,000 users a day and there are 3 million simultaneous users on the network at any given time.
• Skype has been downloaded 163 million times in 225 countries and territories.
• Skype is available in 27 languages.
• Skype has more users and serves more voice minutes than any other Internet voice communications provider.

- Skype Technology

The only pre-requisite for using Skype is that the user must have Skype software running on the computer. This software can be downloaded for free and easily installed.

Skype scores over other VoIP clients as it operates on a P2P model, not the traditional client-server model. These decentralized P2P networks scale indefinitely without increasing the search time and without the need for costly centralized resources. And since the processing and networking power of the end users is utilized, each new node added to the network adds to the potential processing power and bandwidth of the network.

Another point on which Skype scores over other protocols like SIP and H.323 is its ability to permit calls to traverse symmetric NATs and firewalls . SIP and H.323 fail here because they employ User Datagram Protocol (UDP) and point-to-point networking making NAT traversal problematic.

Skype takes care of privacy matters as it automatically encrypts everything before sending it through the Internet. Since all calls are routed through public Internet, encryption is a real necessity. Then on arrival, the message is decrypted on the spot and turned back into high quality voice, plain text or regular file.

In spite of all these features, Skype does face criticism because it has a closed code and the protocol is proprietary. SIP and H.323 are open standard. The use of propreitary protocol makes it difficult for other providers to interact with Skype network.

What’s similar and what’s not between SIP and Skype:

SIP	Skype
SIP operates on a traditional server-client model	Skype operates on peer-to-peer model
SIP has costly centralized infrastructure	The user directory of Skype is decentralised and distributed among the network
NAT and firewall traversal is problematic	Skype allows calls to traverse NAT and firewall
“There are 3 million SIP phones.” [http://www.sipphone.com/]	Skype has millions of download but it is not necessary they translate into actual usage. Disclosure of actual calls made per day will be the actual indication of its usage
The open standards of SIP permit different vendors to build different pieces that will interoperate with others. There is a wide range of hardware like business or consumer phone, adapters, routers and they can be directly plugged to Internet	Skype has a proprietary protocol. It doesn’t have enough hardware to brag about. It must build each piece or commercially motivate a company to do so
All companies can use SIP enabled system to avoid phone bills!	All private users can use Skype to avoid phone bills!

3 – FUTURE OF VOIP

To sum it all up, IP phones conforming to SIP effectively allow IP calls to be automatically routed to compliant handsets.

The technology is able to determine the end system that will be used for communication sessions and automatically sets the appropriate parameters at both ends of communication. It then manages both call transfer and call termination. SIP also allows users to initiate and receive communication from any location.

But the hindrances are the firewall issues, unnecessary complexity, and the cost associated with adopting SIP. Skype technology has an edge here; the biggest plus point for Skype being that there are more Skype users today than SIP.

Today VoIP technology has reached a state where we have the best hardware and the best services. The research behind it and the way the protocol will move forward in future will only make telephony cheaper, that is, unless the big VoIP business tycoons make sure you still pay a premium. With Skype’s acquisition by eBay (somewhere between $3.7 billion to $4.1 billion) free Internet phone calls may not be free anymore! The question that looms today is “Will eBay seek to recoup its huge investment by charging for the service?” [http://www.eweek.com/]

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