(click thumbnail)Darren LevyTieline Technology has introduced 3G hardware plug-in modules for its Commander codec. At the spring NAB convention it spent time with attendees discussing how 3G networks are enabling broadcasters to send wireless stereo FM-quality remotes at bit rates comparable with ISDN performance. The company also has been actively monitoring developments in audio over IP. RW discussed these topics via e-mail with Darren Levy, Tieline’s international marketing manager.
Someone said during the recent NAB that audio over IP will be as important to radio as the development of the Marti was. Why?
Our Kevin Webb says this in reference to wireless remotes. The Marti was the first big wireless remote revolution but it was only in one direction, was mono and only 7 kHz audio and was dependent on line-of-sight thus could not be used outside of one’s coverage area.
This new high-speed wireless capability is literally changing remote broadcasting history for the first time since the Marti. Broadcasters can now do broadcast-quality stereo (or dual mono) audio remotes from anywhere in the country where high-speed wireless data is available back to the studio with no long-distance charges or minutes used (when using an Unlimited Data Plan). No more borrowed phone lines, no ISDN lines to set up, just show up, turn on the Tieline, broadcast the remote and leave.
What developments in audio delivery and wireless bandwidth should U.S. readers know about?
Wireless is the biggest and latest development. Major cellular carriers are in a fierce battle to provide high-speed wireless data. Broadcasters win because this gives them the ability to do “FM-quality” stereo (or dual mono) wireless remotes using Tieline’s high-quality wireless systems.
As cellular carriers continue to improve their wireless data speed and coverage, eventually the majority of U.S. broadcasters will be able to deliver broadcast quality audio remotes from wherever they can get a broadband cellular signal, possibly by the end of this year according to some carrier’s predictions. (Sprint has said 210 million people will have access to higher speed wireless data by the end of 2007, as an example.)
Wired Internet service providers are also offering lower-cost, higher-bandwidth links allowing broadcasters to take advantage by delivering audio over wired IP networks such as the Internet, LANs and WANs.
Telcos are now actively refusing to supply “new ISDN links,” which is forcing broadcasters to look at alternatives methods to transfer audio.
Telcos are also actively converting analog lines between exchanges from traditional POTS routing links to VoIP routing links to enable greater VoIP traffic and respond to the price erosion introduced by Internet telephony. This will ultimately force POTS codec users to begin converting to VoIP and IP-based packet switched codec technologies and wireless remotes will eventually replace most POTS remotes.
Give examples of how your company’s products are evolving to reflect these trends.
Our third-generation G3 open design platform recognizes that telecommunications networks will continue to evolve. By adding relevant network modules to the codec, Tieline customers are able to use existing infrastructure such as POTS and ISDN and transition into new networks such as IP and 3G at a very low cost.
Tieline is the only codec in the market to support most current network transports including wireless broadcast quality audio over 3G broadband wireless cellular networks; IP, LAN, the Internet and BGAN satellite networks; analog telephone lines commonly known as POTS or PSTN links (including leased 3 kHz dry pairs); wireless GSM cell phone networks; ISDN BRI and Inmarsat satellite circuits; and digital leased lines with X.21 and V.35 interfaces.
Is the cellular network “ready for prime time”?
The same question could have been asked about ISDN and POTS networks several years ago as emerging codec technology sought to deliver broadcast quality programming over challengingly low bit rates.
To me, the real question should be: “Are today’s codecs equipped to deal with lossy wireless networks and deliver broadcast quality audio with low latency and stability?” The answer is yes. This is the new telecommunications frontier which all broadcasters face, and this has been Tieline’s challenge for the past two years.
The answer also lies in using more modern algorithms that have been engineered to manage lossy networks plus a delicate combination of network congestion management and forward error correction strategies using the UDP network protocol. Quality of service that prioritizes packets over the link is also advisable wherever possible.
In addition, Cingular, Sprint, Verizon and Alltel are providing broadband data access over their cellular network and continue to improve wireless data speeds and coverage each week. Right now, we have been experiencing anywhere between 64–192 kbps and in some cases up to 256 kbps links over cellular broadband networks. Like ISDN, these speeds are good enough to deliver high-quality stereo audio. However these cellular network speeds are improving all the time.
Traditional algorithms such as G.711, G.722 and MPEG L2/L3 were engineered for circuit switched networks some 20-plus years ago. While wired IP links with QoS will support them, we find that these algorithms perform poorly over wireless IP links that require low to moderate delay over the open Internet. AAC, AMR and Tieline Music were engineered to cope with lossy networks at low bit rates.
For example Tieline Music can deliver 15 kHz stereo at 64 kbps over a wireless cellular broadband network and there are broadcasters all over the globe using this every day.
Whereas circuit switched technologies send each audio byte one after the other in the right order, packet switched networks can route packets in many different directions so that they end up arriving at different times and often in the wrong order. (Some don’t ever turn up.)
Tieline has engineered a range of forward error correction and concealment strategies to ensure seamless audio delivery over wireless networks.
Wireless network congestion occurs when a large number of people suddenly require access to both the voice and data channels in a given cell area. The cell reacts by changing the bandwidth for each user to ensure all users can get access. This is quite different from circuit switched ISDN and POTS networks, where you are guaranteed the bandwidth of your link. To manage this codecs need to produce high-quality audio at low bit rates so that there is enough headroom for network fluctuation without interrupting the broadcast.
At Tieline we built our reputation in the POTS market by specializing in “prime time” quality audio over ultra low bit rate networks.
Tieline has been talking a lot about audio over IP, and in particular its platform for interoperability over IP using SIP. Why is it important to broadcast users?
When ISDN codecs were emerging, most manufacturers designed proprietary ISDN codec hardware and software that was incompatible with other brands. Many large broadcasters were not prepared to rely on a single codec brand for their audio contribution links and they purchased multiple brands.
With ISDN circuits being phased out in the next few years and the transition to IP ahead of them, those same broadcasters (particularly in Europe) clearly stated to codec manufacturers that they were not prepared to go through the same pain as the early ISDN years. They stated that they would not purchase new codecs until they could see a number of brands develop a standard for interoperability.
The European Broadcast Union is an organization dedicated to working with broadcasters and manufacturers to develop global standards for audio and video contribution. The EBU has recently released a draft specification proposing a standard, which includes Session Initiation Protocol.
SIP is an application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution and multimedia conferences.
Most telcos around the world have standardized on SIP to create VoIP connections between devices so it is logical that the broadcast industry follows the specifications of the telcos that we rely on for our broadcast links.
Tieline and a number of codec manufacturers in Europe announced they were implementing SIP to address interoperability concerns and Tieline has successfully tested SIP based connections with Prodys, Aeta and the new Musicam IP-enabled codecs.
Having spoken to most codec manufacturers at NAB2007, I got the impression that each manufacturer was willing to implement SIP to create a basic framework for interoperability. Manufacturers are yet to agree on the choice of algorithms suitable for packet switched networks; however the EBU specification currently proposes G.711, G.722 and MPEG 1/2 Layer II as mandatory for interoperability, and AAC and MPEG 1/2 LIII are recommended; APT and AMRWB are optional.
The European Broadcast Union has issued interoperability standards. What’s happening in the United States? And is it reasonable to think that U.S. codec competitors will agree?
I don’t think American manufacturers are under the same time pressure for interoperability as European manufacturers. For example, in Sweden ISDN will be significantly withdrawn from the market by the end of this year and other European telcos are making similar noises for 2008–2010.
In the American market, Telos has made the AAC suite of algorithm profiles popular amongst broadcasters and the iPod has made it a household name (even though the broadcast and iPod versions are not compatible). This algorithm suite has superior audio performance to the MPEG Suite of algorithms plus it supports profiles for handling lossy networks. It is also capable of working over both high and low bit rate networks and with AACLD, can deliver lower latency than the MPEG Suite.
I believe American manufacturers have a preference for interoperability using AAC. We understand why and we also think this is a good idea.
Therefore at this stage it is not clear how much of the EBU draft proposal will be followed by American manufacturers. …
At NAB I understood American manufacturers to be interested in interoperability using AAC at 32 kbps, 64 kbps and 128 kbps sampling at 48 kHz at a basic level, then adding SIP as a signaling layer.
As Tieline has already implemented SIP with the MPEG Suite of algorithms, we will also support the proposed AAC interoperability.
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