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Special Focus
March 2003

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Setting The Course For Profitability And Growth


While the telecom seas remain rough, there is a growing consensus about the proper course for the telecom industry to take as it attempts to navigate towards the safer waters of sustained profitability and growth.

The course is simple: Develop and deploy new, differentiating enhanced IP services that can drive up both subscriber demand and service provider profits.

This sounds simple, but questions still remain. What are the tools needed to drive these new services? How are these new tools implemented? The answers lie with SIP and a SIP-based services infrastructure.

Session Initiation Protocol, or SIP, has emerged as the control protocol of choice for U.S. and international service providers that are deploying converged services and IP infrastructures. SIP is a simple and flexible call control and signaling protocol that enables the rapid and efficient development and deployment of innovative third-party converged applications in service provider networks.

Developed under the auspices of the IETF, SIP facilitates the setting up, changing, and terminating of multimedia sessions across diverse network infrastructures.

Three key attributes make SIP the new gold standard for enhanced services.

� SIP is ubiquitous. Because SIP is based on established, open protocols (derived primarily from HTTP), there are a large body of open implementations, ensuring a vibrant, competitive set of offerings and ongoing innovation. This openness enables the real time integration of diverse voice, video, and data content (Web, e-mail, etc.) into applications, broadening application features and providing a richer end user experience.

� SIP is simple. It is a text-based protocol that�s easy for developers to understand and use. SIP leverages proven Web development and deployment paradigms. It substantially widens the pool of development talent and available tools. Most importantly, this paradigm makes new service deployment faster, cheaper, and easier to operate.

� SIP is flexible. SIP-based architectures are distributed, much like PC architectures of the IT world. Just as the PC supplanted centralized mainframes, SIP puts the intelligence for call control and features on distributed devices. Such devices include SIP proxies and SIP Application Servers in the network and Media Gateway Controllers (MGCs) and SIP phones or soft phones (PCs) at the edge of the network. This contrasts to the centralized model of the TDM world where processing and control intelligence reside on large phone switches or server in the network core.


SIP works with existing Internet protocols, enabling endpoints to discover one another via network hosts, and agree to share sessions. This application-layer control protocol lets users establish, modify, and terminate multimedia sessions such as conferencing, Internet telephony calls, and instant messaging sessions Overall, SIP supports five functions for establishing and terminating multimedia communications:

� User location -- Identifying end systems to be used for communication.

� User availability -- Determining the called end user�s willingness to engage in a call.

� User capabilities -- Specifying the media and media parameters to be used.

� Session setup -- Establishing session parameters for both the calling and called parties.

� Session management -- Modifying session parameters, invoking services and terminating the session.

SIP conferencing enables the easy integration of new multimedia features and greater end user flexibility and control. What does this mean to the consumer? No longer do conference calls require users to call into a centralized bridge at a set time with no guarantee of 100 percent attendance. Utilizing presence-based technology and a convenient Web interface, a blast to your IM buddy list or e-mail contact list enables a conference call to be arranged in seconds. Callers will no longer be subjected to the painful sounds of elevator music nor have to engage in the mindless pre-conference chit chat while waiting for a conference call to commence.

SIP Conferencing means calls commence in seconds with 100 percent attendance due to presence. Callers can select their own choice of musical accompaniment if they want, by piping in a popular MP3 tune from a Web site and playing it into the call. For business conference calls, new enhanced features such as peer-to-peer file sharing enables conference participants to collaborate in real-time on joint projects such as sales presentations or architectural drawings, all with the click of a mouse. The �hurry up and wait� of TDM-based conferencing is transformed into a real-time, interactive communications experience that is not only more compelling and flexible for the user, but a good deal cheaper.

From the service provider perspective, enhanced SIP conferencing is a far easier application to develop and deploy. It requires an IP-based services infrastructure consisting of a conferencing application server and a media server platform.

� The conferencing application server holds the logic for the application. Its functions include establishing the conference, providing participants with the information necessary to join the conference and all aspects of conference control and management. The conferencing application server uses SIP to control the media server and to manage and direct media sessions during the conference call.

� The media server works on behalf of the conferencing application server and receives directions via SIP from the application server. The media server mixes media sessions, delivers audio streams, plays and records media to and from the conference and supports all the IVR functions required for the conference on behalf of the application server. The media server also has the capability to perform transcoding functions for multimedia terminals.

For the service provider, the deployment and management of a SIP conferencing solution gives them more flexibility and efficiency and most importantly more profit potential.

Although vendors and service providers still must weather the telecom storm, the course has been set. SIP and SIP services offer unprecedented opportunity. SIP�s ubiquity, simplicity, and flexibility make it a compelling technology for navigating your way to the calm and safe waters of profitability and growth.

Eric Burger is CTO and co-founder of SnowShore Networks. He is a key contributor serving on numerous working groups for SIP and VoiceXML for the W3C, IETF and 3GPP. Burger co-authored the SIP application media server interface requirements draft for the IETF and has guided the development of VoiceXML for the W3C. For more information, visit the company�s Web site at www.snowshore.com.


The SIP Center is a portal for the commercial development of the Session Initiation Protocol. Serving both the SIP community and the wider industry, the SIP Center offers comprehensive technical and market resources as well as an environment for the testing of SIP implementations. Visit The SIP Center online at www.sipcenter.com.


[ Return To The March 2003 Table Of Contents ]

SIP Conferencing Call Flow

� Set Up: Conferences are created dynamically by sending an INVITE command from the conferencing application server via media server out to the conference callers to initiate the conferencing session. The conferencing application server knows the participants are available and willing to participate in the conference from their presence information.

� The first call leg to the media server serves as the conference control channel for future changes to the conference provisioning and serves as the gateway to play/record media to and from the entire conference.

� Adding Callers: Participants join the conference by accepting the SIP INVITE message from the conference application server. The application server then issues a SIP INVITE to the media server to join the callers into the conference mix.

� Modifying the Conference: Changes to the mixing mode or event subscriptions are achieved by issuing a re-INVITE or INFO on the selected call leg with a MSCML message body indicating the desired conference leg parameters.

� Removing participants from a conference: The application server removes a participant from the conference by issuing a SIP BYE command that deletes that participant.

� Playing Audio: Audio may be played into the conference by issuing a re-INVITE on the conference control leg. In the re-INVITE message is a message body specifying the audio to be played and related parameters.

� Leaving the Conference: A participant can always leave the conference by issuing a BYE command to the conference application server. The conference application server will then issue a BYE command to the media server for that leg. Likewise, the conference application server can eject a participant by issuing a BYE command. So long as the conference control channel is active, the conference is active at the media server. This allows for conferences to exist even if all of the participants drop off of the conference momentarily.

� Ending a Conference at the Media Server: The conference application server issues a BYE command to the media server on the conference control channel. If there are remaining participants at the media server, the media server will issue BYE commands on those sessions.

[ Return To The March 2003 Table Of Contents ]

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