Tieline Technologies

Interoperability over IP with other codec brands using SIP

Apr 1, 2007

NAB Las Vegas  (April, 2007) - Tieline Technology, a leading provider of high-quality remote broadcast digital audio codecs, will demonstrate codecs that are compliant with the new audio-over-IP interoperability standards issued by the European Broadcast Union.

"We see the new interoperability standards as a positive step for our clients around the globe," said Tieline America General Manager Kevin Webb. "In the end it will lower the cost of ownership for broadcasters and lower their frustration level."

The scaling down and in some cases the complete shut down of ISDN networks in the USA and Europe is prompting the EBU to create operability standards that will allow IP codecs from different manufacturers to communicate with each other.

"ISDN codec end users have enjoyed interoperability between most codec brands over the past 10 years," said International Marketing Manager Darren Levy. "As broadcasters are now being forced to migrate to IP networks they are looking to the EBU and manufacturers to establish a global standard for reliable broadcasting over IP networks and interoperability between brands over these new networks."

Among a package of standards being tested:

  • The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet.
  • Algorithms: mandatory, recommended and optional
  • SIP or Session Initiation Protocol -- is an "Internet" application-layer protocol that allows audio and video sessions to be established over the internet.  SIP can work seamlessly with existing telecommunication service providers allowing audio and video to be exchanged across different network types.  For example, SIP is often used to allow internet VoIP phones to connect to existing PSTN phone networks. SIP provides many features to simplify a codec connection across the internet.  These include:
  • The ability to register a name or number to a codec to simply dialing via a SIP Server.
  • A mechanism to allow the best algorithm that is common to both codecs to be used.
  • Correctly setting sample rate and bit rate as well as Mono/stereo mode
  • Option to send only, receive only or standard send/receive mode.

Tieline plans to extend its support for phone voice mode to include SIP VoIP over IP connections.  Tieline now offers Voice mode connectivity on POTS, GSM, ISDN and IP.
 
"SIP is promising because of its natural integration in the IP world, its ability to evolve and its flexibility," said Webb.

SIP is a key enabler for successful services such as instant messaging, as well as new services such as VoIP. SIP has also been selected by the 3GPP (Third Generation Partnership Project) as a major component of IMS (IP Multimedia Subsystem) for the evolved UMTS core network.

"A key for broadcasters is that implementation of SIP allows for remote broadcast via IP without having to acquire a dedicated IP address. Broadcasters will be able to simply dial a phone number," said Webb.

Tieline codecs featuring SIP interoperability will be demonstrated at the Tieline booth at NAB2007 (N9311).    

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