Protocol Development

In the last ten years, the number of users on the Internet has increased at a dramatic rate with more and more applications deployed in the data world and no end in sight. The intent of one of these applications was to carry voice over a data network; this eventually took the name of Voice over Internet Protocol (VoIP). The initial VoIP deployments had two functions: to convert voice from/to data packets and es-tablish a protocol to get the phones in contact with each other, similar to the switching concept in the TDM world.

As IP deployment continued its explosive growth, VoIP became increasingly attractive, and the need for standards became obvious. With-out standards there was no hope of interoperability and user features were limited to the basic Plain-Old-Telephone-Service (POTS). Also, administration of larger networks and security would become a nightmare.

One of the first efforts at standardization resulted in the H.323 standard. H.323 is really an umbrella specification encompassing many addi-tional specifications. The "H.xxx" signifies that this specification comes from ITU-T (formally CCITT). H.323 may be efficient in terms of memory and CPU usage, but it is difficult for programmers to work with, and difficult to expand and develop. REDCOM, and the VoIP industry in gen-eral, have embraced SIP as the best protocol for converged communication going forward based in ease of use, human readability and SIP's ability to expand.

SIP Capabilities

Session Initiation Protocol (SIP), in contrast to H.323, represents a complete break with the tradition of TDM telephony protocols as it comes from the data world. SIP is not a specific protocol for telephony. Rather, SIP is a generalized protocol for allowing what are called "user agent" clients and servers to associate, and when communication is desired, allow them to exchange capabilities, make media choices and establish communication sessions between them. SIP can mediate a session involving a one-way transmission of a full-length movie, a multi-party video conference, a telephone call, or an Instant Message transmission. SIP provides a set of methods and a suggested way to use them to create (in the case of VoIP) call flows for most of the commonly used features we are familiar with in the TDM world. Just a few years ago, SIP was not rich enough to permit more than simple POTS telephone service without giving up interoperability. For example, VoIP phones could not be mixed and matched.

There are almost daily improvements being made in the SIP implementations as more and more vendors, including REDCOM, follow SIP standards. This increases interoperability between different vendor equipment.

TRANSip Features

As the convergence of TDM and IP networks continue, end users do not necessarily want to abandon their traditional telephony features.

TRANSip's integrated architecture allows the migration of many rich traditional TDM features to standard SIP phones, but certain specific fea-tures require support from the phone manufacturer (i.e. support for hook-flash). The following is a partial list of traditional TDM features that are supported by TRANSip.

Selected Features Available to SIP phones

Compatible SIP Phones

In the traditional TDM world, the phone or the end station was easily compatible with the existing phone lines. The backward compatibility is so strong that an operational phone from the early 20th Century can be connected to the existing telephone network.

As the intelligence shifted to the end terminals in the IP world, the phones started to deploy new capabilities. However there are still inter-operability issues as the phone vendors may deploy proprietary signaling protocols. In order to ease the concern of its customers, REDCOM periodically tests various phones in the industry.

Below is a list of IP end devices that have been tested by REDCOM for compatibility with TRANSip as of August 2010: