Delivering IP-Based Voice Services with SIP
By Tom Baumgartner, Director of Product Marketing

NMS Communications - SIP Center Principal Sponsor  SIP Center Principal Sponsor

A recent survey of 300 business technology executives1 shows 29% use VoIP, 18% are testing it, and 34% have plans to deploy VoIP - so there is no doubt that VoIP is making its mark. Key to the successful implementation of any voice over IP solution is a call control protocol that is widely accepted, easy-to-use, and provides all the user features demanded of today's telecommunications services. While the H.323 ITU-T standard for the transmission of real-time audio, video, and data information over packet-switching networks currently commands a large installed base, mainly because it existed when VoIP first emerged in the mid 1990s, the Session Initiation Protocol (SIP) from the Internet Engineering Task Force (IETF) is now in contention for leadership.

In recent years there has been a lot of innovative work done to define clarifications, bug fixes, tweaks, and extensions to SIP. Notable examples include the IETF clarifications of procedures for transferring calls, and new requests for comment (RFCs) and drafts to support the administration and attendant features common to PBXs and call centers. Recognized problems for SIP, namely operation through firewalls and Network Address Translation (NAT), have been address via RFCs and drafts. In addition a body of literature is now available to describe how to use SIP, the Session Description Protocol (SDP), and other IETF standards to provide every imaginable voice service over an IP network.

The downside of all this progress for developers is the continued fluidity of the SIP standards. Successful products must sometimes implement drafts in anticipation of the final standards. But the eventual dominance of SIP as the VoIP call control method of choice seems inevitable. The SIP advantages - scalability and interoperability - are derived from the architecture of the Internet that has intelligence located at the edge of the network (no central control) and devices communicating using standard protocols.

SIP Infrastructure

Figure 1 shows the components of a SIP infrastructure. User Agents are end point devices such as SIP telephones. User Agents originate SIP requests and send and received media. The function of the various SIP servers is apparent from their names. The Registrar Server accepts registration requests from User Agents and updates the User Agent information in a Location Service or other database. The Proxy Server receives SIP requests from User Agents or other proxies and forwards the requests to another location. The Redirect Server receives requests from User Agents or proxies and returns a redirection response, indicating where the requests should be retried. The Presence Server accepts subscription requests and notifies the subscriber of the presence of a User Agent. This function has a number of uses, but the most common may be to facilitate Instant Messaging.


Figure 1. SIP Infrastructure Components

SIP Methods and Responses

The SIP protocol defines a number of request messages, known as methods and responses. For example, the INVITE method requests the establishment of a session for passing media. Other common methods are BYE (session termination) REGISTER, REFER (pointing to another address), SUBSCRIBE, and NOTIFY. Responses in SIP are numerical. The basic set of messages to establish a session is shown in Figure 2.

INVITE
100 Trying
180 ringing
200 OK
ACK

Figure 2. Successful Establishment of a Media Session

Recent Market Developments

In the last year Nortel, Avaya, Mitel, NEC, Siemens, and Alcatel have released SIP strategies and products. This represents a major disruption in technology for these established PBX and switch manufacturers that have huge investments in proprietary equipment. Most have employed a SIP server called a Back-to-Back User Agent (B2BUA) to act as a gateway between their proprietary protocol and the standards-based SIP environment. It remains to be seen if this strategy will prevail in the long term, but the migration to SIP is well under way.

Voice over IP represents the future of telecommunications and SIP is the control protocol that will enable that transition. Next month’s newsletter will describe the SIP offering to be released in Natural Access 2005-1 that will help developers get their SIP-based solutions to market quickly and cost effectively.

To Learn More….

To learn more about SIP, check out these web sites:

Tom Baumgartner may be reached at + 1 847 925 8900, ext. 5029 or tom_baumgartner@nmss.com.

1Information Week, March 1, 2004, page 36

Source:
http://www.nmscommunications.com/News/TelecomInnovators.html#Feature1