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This is part two of a set of articles describing the benefits of Aculab’s range of IP-centric Prosody X media processing boards for developers of large scale communications platforms.
In part one, it was stated that there are many advantages in having IP at the core of the Prosody X media processing board, not least of which is the high channel counts made possible by such an architecture. But that is just one aspect…

Choosing IP connectivity enables a distributed architecture

In putting IP technology at the heart of the Prosody X design it is much simpler, due to the ubiquity of Ethernet in IT networks, to build multi-board, multi-chassis and multi-location systems using Prosody X. So if the high channel counts of a single Prosody X board are not enough, then it is easy to build multi-board systems.

The traditional way

Before VoIP entered the telephony world, we depended on TDM technology and an array of disparate CAS and CCS type PSTN protocols, and most of the voice-based solutions had a common architecture, simply described as a host server with telephony boards.

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Dealing with RTP loss

September 14, 2011 10:11 AM | 0 Comments
In my last post, you read about SIP call recovery from RTP loss. This post continues the theme, providing some more useful information – detecting and dealing with RTP loss between two SIP end points.

If you recall the scenario, there is a gateway installed between an SS7 carrier network and an emergency services IP network – an ESInet – over which emergency calls are routed to public safety answering points (PSAPs). Calls entering the ESInet via the gateway are SIP/RTP/UDP and, as the NENA i3 specification quite rightly states, in no circumstances should an in-progress emergency call be taken down automatically, just because RTP streams fail.

Whatever effort you put into making the call handling equipment redundant and resilient, it is always best to assume that bad things can still happen. To misappropriate the principle of simplicity from Ockham’s razor, follow the rule of thumb that advises, “Make as few assumptions as possible.” In fact, make one assumption only – expect the worst (i.e., ‘the proverbial’ happens).

So, we can expect situations to arise in which RTP media between the gateway and the ESInet fails.

PSAP equipment must be able to distinguish between RTP failure and real silence by a caller. Continue Reading...

SIP call recovery in gateways

September 2, 2011 11:08 AM | 0 Comments
Life is full of truisms; those self-evident truths in statements like not being able to recover your lost youth (despite what plastic surgeons say – or any number of elderly film stars have wished). It’s not as evident that you can’t recover SIP (session initiation protocol) calls when they appear to fail. What happens when you lose an already established SIP call between a gateway and an end point on a VoIP (voice over Internet protocol) network? Can it be recovered?
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The team here at Aculab are gearing up for NENA 2010, which takes place in Indianapolis, USA in early June. NENA 2010 is an event run by the National Emergency Number Association (NENA) for public safety professionals, telecoms specialists and government officials, which focuses on the near and long term issues facing public safety in the USA.

NENA is calling for the migration of 911 networks to next generation E911. The idea is to move 911 systems to standards-based IP platforms and, in the process, enable citizens and those involved in emergency response to interact not only in voice, but also via text, IM and possibly even video communications.

The ultimate goal is that citizens, no matter who they are, where they are or which communications device they use, can make an emergency call and the emergency services will have the accurate location and subscriber information to find them.
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Is the lack of widespread support for full 'five-9's' reliability in IP-based networks  holding back the adoption of IP communications?

It is certainly something we thought might be the case, and so we set our engineers the task of providing redundancy mechanisms for IP communications , based on Aculab's SIP stack, that would enable end customers to build in the reliability they were craving.

Our goal was to enable our customers to build enterprise and service provider communications platforms that could match the reliability and resilience of a legacy TDM-based system, but also give the full unified communications experience made possible by IP.
99_999% image.PNG
For over 30 years, designers of TDM networks have taken every step possible to minimise network downtime in search of the mythical 99.999% availability figure (just over 5 minutes downtime per year). The basic rule has been to eliminate any single point of failure. This philosophy has extended beyond simply duplicating the network hardware - separate buildings with separate power feeds and physical connecting cables are also a key part of a resilient network design.

For the IP-core network consisting of routers that are inherently designed to utilise multiple routing paths, this is not an issue; but for hardware attached at the network edge such as media processing servers, it is not so straightforward. An IP-based media server supporting VoIP and video services and using SIP for its signalling is one such platform that sits at the edge of the service delivery network.
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Recent emergency situations, such as the earthquakes in Haiti and this week in China bring it home to us that when disasters like these occur, the emergency services and the general public who are directly affected rely heavily upon communication systems to co-ordinate rescue efforts and receive the most up-to-date information.

As a supplier of enabling technology that often gets installed into the communication systems used by emergency services, Aculab has a duty to enable such systems to operate flawlessly, to provide communications clarity and to offer utmost reliability.

Several recent technological advances are playing their part in providing world-class Integrated Command and Control Systems (ICCS):

The first example is a technique we have developed to enable the concept of equipment duplication used in TDM communications systems to give 'five 9s' reliability in an IP-based communication system. The Dual Redundant SIP Service (DRSS) enables the SIP stack used for the call control to be duplicated over separate hardware platforms to give much improved fault tolerance capabilities. By keeping the two SIP stacks synchronised, a call being set up or in progress will not  be dropped even if the main server fails mid-call or during call set-up.

Another example where new technology will give advantages to emergency communications system developers is in the adoption of video codecs and wideband audio (HD voice) codecs on communication platforms. Continue Reading...

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