Audio over IP…Are you ready for the Remote Broadcast revolution?

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Click the above TABS for more detailed information on the Features and how Audio over IP works

Features

 

Features of Tieline IP Solutions
Tieline IP enabled codecs can reduce the costs of broadcasting by facilitating the migration from increasingly scarce and costly fixed line synchronous networks, into low cost packet-based networks.

IP connections are increasingly widespread, relatively cheap to install and they provide scalability over time. They also provide the opportunity for multipoint routing of broadcast signals.

• Connection flexibility with a range of algorithms offering broadcast quality 7.5kHz mono audio at bitrates as low as 9.6Kbps, 15kHz mono at 24Kbps, 15kHz stereo at 56Kbps and up to 20kHz stereo at 96Kbps.
• Algorithms optimally suited to IP broadcasting include Tieline Voice G3, Tieline Music and Tieline MusicPLUS. Other algorithms available include G.711, G.722 and MPEG Layer 2.
• Tieline’s ‘QoS Performance Engine Technology’ software significantly enhances connection reliability over lossy IP networks such as 3G cell-phone networks and the Internet.
• Configurable Forward Error Correction (FEC) with automatic and adaptive jitter buffering, allows codecs to withstand significant packet loss over IP and 3GIP connections and still deliver continuous high quality audio, with very low delay.
• Tieline’s extremely high quality audio provides compatibility with HD radio broadcast chains and is ideal for Internet broadcasting.
• SIP compatibility provides interoperability with most other brands of codecs.
• One touch dialing from the field provides automated dialing for inexperienced users and minimizes the need for technical staff at remote broadcast sites.
• Remote control of your talent’s audio input level from the studio means you receive optimal levels without the need for technical support on site.
• Automatic configuration of a codec’s IP address, subnet mask and default gateway from DHCP networks, allowing simple connection of a remote codec to a wired DSL or Wi-Fi hotspot for dialing back to the studio.
• Manual configuration of the IP address, gateway, subnet and ports is also possible for proprietary and highly secure networks.
• Uncompressed (Linear) 15-23kHz audio - suitable for high quality Studio-to-Transmitter links (STL), Studio-to-Studio links and high-end applications.
• Auto configuration of the remote codec from the dialing codec (i.e. the remote codec will recognize the settings you have created in the dialing codec and be automatically configured).
• Optional Automatic failover to a POTS telephone line, ISDN, Satellite, X.21 or GSM.
• Simultaneous broadcasting over IP and your choice of POTS, ISDN or GSM.

How IP works

 

An Introduction to IP
Tieline codecs use simple wizards to program IP connections and can be programmed to connect at the touch of a button. Notwithstanding this, it is helpful to have some understanding of how IP networks work.

Using IP you can connect codecs over a private Local Area Network (LAN), or over different public WAN networks such as the Internet. You can also use a 3G/3.5G cell-phone network to connect wirelessly over IP. Codecs can connect via hard-wired Ethernet connections or a combination of hard-wired connections and wireless technology.

IP and Data Packet
Circuit switching, as used in ISDN and GSM CSD and HSCSD connections, creates a dedicated connection between two nodes to send data exclusively between two devices.

Packet switching, as used in computer networks, IP codecs and other telecommunications devices (i.e. 3G cell-phones), is where data packets can be individually routed between two nodes (in our case two codecs) over either dedicated (e.g. STLs) or shared (e.g. remote broadcast) LAN or WAN connections.

Packet switching optimizes the use of bandwidth over computer and wireless networks by dividing data streams into packets with destination addresses embedded within them.

Sometimes packets arrive late or are lost over lossy 3G/3.5G cell-phone networks and the Internet. Tieline’s QoS Performance Engine combines features such as Forward Error Correction (FEC) and automatic jitter buffering, to withstand significant packet loss over IP and 3GIP connections and still deliver continuous high quality audio

Latency – Delay over Packet Switched Networks
Latency, or delay, is the amount of time it takes for a packet of data to get from one point to another. Over packet-switched networks delay is variable, depending on the path packets take from their source to their destination. Latency is an important issue when using packet-switched networks – particularly when broadcasting audio or video in live situations. Latency over packet-switched networks is created by:

1. Network transmission delay;
2. Physical processing delay over the network via switchers and routers etc.;
3. Packet delay including algorithm compression delays.

All of these factors contribute to the total latency over a network. If the total latency over a network is too high then it may be difficult to sync up with other audio and/or video sources when conducting interviews etc.

Tieline’s automated jitter buffering solution eliminates issues created by packets arriving in varying intervals over IP networks. It automatically measures network latency to minimize program delay and provide peace of mind when streaming audio over IP and 3G/3.5G cell-phone networks.

More Detailed Info on IP
Tieline’s ‘IP & 3GIP Streaming Reference Manual’ details all the information you need to know, and more, in relation to broadcasting over IP. Some sections you may like to read include those relating to:

1. Public versus Private IP addresses: What are the differences?
2. Jitter buffering: Dealing with latency over IP networks.
3. Forward Error Correction (FEC): Dealing with packet-loss over IP networks.

In addition, the codec quick-start guides in our support sections contain information about connecting quickly over IP and 3G/3.5G IP networks and tips about how to get the best connection over IP networks.

Download the ‘IP & 3GIP Streaming Reference Manual’

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