IP Audio Overview
Audio over IP has proved itself to
be the broadcast network infrastructure for radio and television,
both today and into the future. As a consequence, increasing
numbers of broadcasters are migrating to low-cost wired and
wireless audio IP networks instead of older and more costly fixed
linetechnologies such as ISDN, X.21 and POTS.
IP audio networks are more flexible, cheaper to upgrade and just as reliable as older network technologies. As a result, broadcasters using Tieline IP audio codecs are able to design and operate more adaptable audio networks with streamlined workflows and reduced operating costs.
Hear the Tieline SmartStream Difference...
Tieline's exclusive and proprietary SmartStream IP technology responds to variable IP network conditions with agility and flexibility to significantly improve connection reliability over lossy IP networks such as 3G cell-phone networks and the internet, especially in those situations when Quality of Service (QoS) is unavailable.
Tieline has built a significant knowledge base from this experience and expertise in broadcasting audio over unmanaged IP Networks. Now SmartStream IP technology takes broadcasting over unmanaged IP audio networks to the next level for uncompromised AoIP.
All Tieline IP audio codecs are EBU N/ACIP Tech 3326 compatible and the company is committed to developing new IP and 3G audio codec applications that take advantage of emerging network infrastructures around the globe.
Tieline IP audio codecs are ideal for use in remote outside broadcasts, as studio-to-transmitter links (STLs) or for studio-to-studio audio distribution. Whether you are sending audio from fixed installations or over wireless remotes, Tieline has an IP solution for every broadcast situation.
Using IP Audio for Remote Broadcasting
Why you should choose Tieline IP Audio Codecs...
Tieline codecs offer a growth path
for each broadcaster's evolving needs, as well as investment
protection against obsolescence. With Tieline you can add to each
codec's capability over time with low-cost plug-in modules for each
connection transport, compared with buying additional codecs.
Tieline IP can be used over DSL/ASDL, cable modems, high speed data cell-phones and satellite IP networks. They can also provide 20kHz uncompressed stereo audio over IP.
Existing Tieline codecs can easily be upgraded for use over IP with a simple software upgrade - and of course you can still use them over other networks such as POTS/PSTN, ISDN, GSM, 3G and X.21. Only Tieline offers you:
- Automatic failover with complete redundancy over multiple network types and the ability to have connections simultaneously dialed up and streaming audio.
- Full remote control of IP audio connected codecs from the studio.
- One touch dialing for connecting over IP and 3GIP networks
- Loss-tolerant Tieline Music and MusicPLUS algorithms that provide stable, high quality connections, at very low bit-rates.
STL over IP Example
It is not unusual for wired internet connections to experience up to 3% packet loss, while wireless IP audio connections can experience 10% or more! With SmartStream features such as Forward Error Correction (FEC) and automated jitter buffering, Tieline IP audio codecs can withstand significant packet loss over wired and wireless IP connections and maintain continuous high quality audio.
Future-Proof your Investment in IP with SIP
Tieline and other members of the
Audio-via-IP Experts Group have taken a leadership role in
developing SIP (Session Initiation Protocol) interoperability
between Tieline IP audio codecs and other codec
SIP makes it a lot easier to connect between different brands of codecs and hugely simplifies remote broadcasting radio and television audio over IP. Using SIP there is no need for IP audio codecs to acquire static IP addresses in order to connect - even if they move between different locations.
Tieline IP Audio Codec Compatibility
Tieline IP audio codecs are the
only ones to support compatibility with other brands of codecs over
five different network types - IP, 3GIP, POTS, ISDN, and
Over IP Tieline is compatible with all major brands of IP codecs that have implemented EBU specifications for interoperability over IP using SIP (EBU N/ACIP Tech 3326). As a member of the Audio-via-IP Experts Group, Tieline is compatible with fellow members Orban CRL, Mayah Communications and AETA, as well as Prodys, Telos, AEQ, AVT, and Digigram.
When connecting over ISDN and X.21, Tieline is compatible with all major brands of codecs using the popular G.722 and MPEG algorithms. Tieline codecs are also compatible over POTS with Comrex Matrix, Access, Vector and Blue codecs.
This means that if you have a Tieline rack unit codec in your studio with a LAN connection attached and a POTS and ISDN module installed, you can accept calls from any ISDN codec, most IP audio codecs, most Comrex codecs and Tieline wireless 3G codecs - now that's compatible!
In fact, with a Tieline rack-mount IP audio codec in your studio you can connect to any brand over ISDN and IP, Comrex codecs over POTS and IP, and receive wireless 3G broadband and GSM connections from other Tieline codecs.
To find out more about how you can integrate IP audio into your portfolio of broadcast options, click on the options on this page. If you're interested in trialing a G3 codec over IP, click here to contact an authorized dealer and request a free demonstration.
Tieline IP audio 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.
- 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
An Introduction to
Tieline IP audio 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 audio networks work.
Using IP you can connect audio codecs over a private Local Area Network (LAN), or over different public WAN networks such as the internet. You can also use a 3G cell-phone network to connect wirelessly over IP. IP audio codecs can connect via hard-wired Ethernet connections or a combination of hard-wired connections and wireless technology.
IP and Data
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 audio 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 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 IP audio.
Latency - Delay over Packet
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:
- Network transmission delay;
- Physical processing delay over the network via switchers and routers etc.;
- 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 cell-phone networks.
More Detailed Info on IP Audio for Radio and Television Broadcasters
Tieline has committed considerable resources to educating television and radio broadcasters about IP audio generally, broadcasting IP audio over wireless IP audio technologies, as well as how to reliably send internet IP audio over Ethernet. See www.ipaudioexpert.com for a complete guide to sending audio over IP, IP audio codec technology and how to become an instant expert in IP audio broadcasting.
Tieline's ‘IP & 3GIP Streaming Reference Manual' details all the information you need to know, and more, in relation to broadcasting audio over IP using Tieline codecs specifically. Some sections you may like to read include those relating to:
- Public versus Private IP addresses: What are the differences?
- Jitter buffering: Dealing with latency over IP networks.
- 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.