Protocol Development SIP Capabilities TRANSip Features Compatible SIP Phones Access Technologies
In recent 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 establish 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. Without 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 additional 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 general, 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.
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.
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.
As the convergence of TDM and IP networks continue, end users do not necessarily want to abandon their traditional telephony features and needlessly throw away invested dollars.
TRANSip’s integrated architecture allows the migration of many rich traditional TDM features to standard SIP phones, but certain specific features 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.
Lineside Features: Users of SIP phones desire the same lineside features that have been available on traditional TDM and analog telephones. At the same time, VoIP offers new functionality that can deliver new features which are not available on traditional telephone systems. The Internet Engineering Task Force (IETF) has proposed SIPPING standards as an industry-standard way to offer lineside features to SIP phones. TRANSip embraces SIPPING functionality for lineside features (such as call transfer, call hold, and 3-way calling).
Line Groups: SIP phones, along with analog and digital phones, can be put in a line group to benefit from group call pick-up, broadcast ringing, speed dial, and many other features.
Station Number Playback: SIP phone users dial a code and receive an announcement that relays the number of the phone they are dialing from.
Caller ID and Name: Caller ID and name features available to the analog and digital phones are available to the SIP phones along with the ability to choose not sending Caller ID and name, as well as blocking calls based on Caller ID.
Announcements: Common announcements can be sent to analog, digital and even SIP phones. This ensures common announcements for all of the users.
Call Detail Records: Traditional call detail records (CDR) that were logged for billing or quality assurance reasons are also available for SIP phones. TRANSip provides records including callED number, callING number, call time, call duration and other detailed information for SIP phones as well as analog and ISDN 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 interoperability 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.
For an up-to-date list of IP end devices that have been tested by REDCOM for compatibility with TRANSip please visit: http://www.redcom.com/products/tested-devices
In the traditional TDM world, subscriber access was relatively straightforward. Phones would be wired to a switch at a central location. Compatibility between traditional switching platforms and traditional analog telephones was generally strong, thanks to the maturity of the technology.
As traditional telephony systems grew in size, it was realized that offering many lines at a central location could become very expensive. This often involved a significant investment in cable plant and switching hardware. A new technology was developed to aggregate lineside connectivity at a location closer to subscribers, to reduce investment in cabling to subscribers. This new technology (Digital Loop Carrier / DLC or Remote) would handle lineside functionality but would still rely upon a central switching platform for call control. Cabling between the DLC / Remote (T1/E1) was significantly reduced compared to individual cable pairs for each subscriber, and could be copper or fiber. GR-303 and V5.2 are examples of industry-standard protocols used to aggregate subscribers on a DLC / Remote, and are both supported by REDCOM.
Over time, consumer demand for high-speed internet service grew significantly. A new technology, Digital Subscriber Line (DSL) was developed to allow high-speed data service over existing telephone lines. This was often provided by a DSL Access Multiplier (DSLAM) located with the central switching platform. As demand for DSL grew, DLC / Remote manufacturers began to integrate DSL functionality into their products, thus eliminating the need for a separate DSLAM. These products are known as Multi-Service Access Node/Platform (MSAN / MSAP) or Broadband Loop Carrier (BLC).
Most manufacturers of MSAN / MSAPs, BLCs, DLCs and Remotes now offer support for VoIP connectivity between the Central switching platform and the Remote. VoIP introduces both benefits and challenges when it comes to access technology. VoIP enabled remotes require only an IP path from the central switching location. Efficiencies can be gained by consolidating both voice and data on to one link, however there are many factors that must be considered in advance, such as: switch support for VoIP remotes, bandwidth, QoS and latency.
REDCOM brings experience in both IP networking and telephony to help build the proper solution to meet your needs. REDCOM is routinely testing new access equipment including IP Phones, Analog Telephone Adapters (ATAs), and Remotes for compatibility with TRANSip. For an up-to-date list of devices that have been tested by REDCOM for compatibility with TRANSip please visit: http://www.redcom.com/products/tested-devices