Welcome to the iPort
The Telos iPort houses eight stereo MPEG codecs in a single 2RU box, capable of either 8 bi-directional or 16 encode-only channels. It can be used for any application where MPEG encoding and/or decoding is
needed for transmission over IP channels such as VPNs, satellite links, Ethernet radio systems, and Telco or ISP-provided QoS-controlled IP services. Applications include studio-to-transmitter links, network distribution
systems, and links to remote studios. With an appropriate server, the iPort can be used for Internet streaming, broadcasting to mobile phones, and audio distribution systems.
The iPort uses the
Livewire standard. A single Ethernet cable is all that's needed for all inputs and outputs plus remote control. If you aren't
using Livewire yet, just pair the Telos iPort with an
Axia AES/EBU or
Analog Audio node to make
a standalone high-density audio codec package.
It uses state-of-the-art MPEG codec technology to conserve network bandwidth, while preserving high audio quality. A range of codec types and bitrates are supported. They're all licensed from Fraunhofer IIS,
the inventor of MP3 and co-inventor of AAC. They are the highest-possible quality implementations, running on a powerful Intel floating-point processor.
Uncompressed audio is transferred via an Ethernet connection to/from your Livewire network, eliminating expensive, space-consuming converters and connectors. A second Ethernet connector interfaces the compressed audio to a WAN.
Configuration and monitoring is via Web, with the front-panel LCD offering immediate local display of status and basic parameters, such as the assigned IP number.
||Mono: 32 kbps
||Stereo: 48 - 320 kbps
||Stereo: 24 - 96kbps
||High-efficiency AAC, also called AAC+
||Stereo: 24 - 56kbps
||Latest generation of AAC-HE with parametric stereo enhancements
||Mono: 64 kbps
||Stereo: 96 - 320 kbps
| Layer 3 (MP3)
||Mono: 64 kbps
||Stereo: 96 -320 kbps
Link Operating Mode
In the link operating mode, the iPort is intended to be used over IP connections that have reasonably good Quality of Service, with controlled packet loss, jitter, and bandwidth. The AAC codec has a
concealment mechanism to deal with occasional packet loss, but it is not intended for conditions where packet loss is routine such as on the public Internet. There are configurable buffers to accommodate jitter, but longer
buffer lengths add to delay. When you need to use non-QoS-controlled links, the
family of products are a perfect solution because they offer adaptive mechanisms to deal
with network problems.
When used with an appropriate server, the iPort can be used as a simple, reliable encoder for Internet or internal audio distribution. Because the iPort generates standards-based MPEG streams, a wide variety
of PC-hosted and hardware players can be used for listening. The iPort is an encoder that can generate multiple output streams. However, to feed a large number of listeners, a server is required. SHOUTcast (yes, that is
the way it is spelled), Steamcast, or other SHOUTcast protocol compatible servers are commonly used for this application. For delivering audio over a LAN or private network, the encoder and server can be together in the
same rack. For public streaming over the Internet, the encoder typically runs in the place where the audio is generated (e.g. a studio) and the server is in a place where a lot of bandwidth is available such as an Internet
Streams that are served by SHOUTcast protocol servers can be heard on Winamp, Apple iTunes, XMMS, VLC, Foobar, MS Windows Media Player, and many other PC software players. Many hardware players are also on the market, such as Logitech's Slim Player and Freecomís network player.
The diagram shows a simple set-up. A Livewire node provides live audio input, which connects directly to the Livewire Ethernet jack on the iPort. The iPortís WAN jack goes to a LAN that connects to a link that leads to a remotely located server, which provides streams to listeners over the Internet. Many variants are possible. For example, a Livewire PC driver could provide the source audio rather than a hardware node. (However, there must be a hardware node somewhere on the network to provide the timing clock.) An installation which already has Livewire equipment would not need a node dedicated to the iPort - you would just select the channels you want to stream directly from the Livewire network.
The original idea for the SHOUTcast server was to accept an audio input from the Winamp player, which has a special "DSP" plug-in. However, since Nullsoft (the company that designed both Winamp and SHOUTcast) published the specifications for the interface, it was established that encoders like the iPort can connect as well.
If you are planning to make public broadcasts via the Internet and you donít want the hassle of running your own server, or donít have enough network bandwidth, you can host your broadcast through a third party that will handle the streaming for you. You can search the Internet for the streaming host solution that fits your needs. The SHOUTcast forums would be a good place to start.
The installation would be like the one in the diagram. In this case, one LAN ties together the node, iPort and an Omnia dynamics processor. Another network links to the WAN (presumably with a firewall in the picture). This offers good security since the LW network is isolated from the WAN. It would also be possible to use a single network on the studio side. In this case, the firewall would be responsible for protecting the LW network.
Omnia 8x is a perfect companion to the iPort. You get 8 channels of high-quality dynamics processing at low cost, and with the same simple one-RJ-45 connection advantage that the iPort offers.
The iPort is an innovative approach to providing MPEG codec functionality by using Livewire for its audio interface. Its features include:
- High density, 8x8 bi-directional or 16 encode-only channels in a 2U box.
- Low cost. Weíve built the iPort on a single industrial motherboard, rather than the usual multiple DSP cards in a frame approach. Together with the Livewire-only audio interface, the iPort costs a fraction of legacy
- Direct livewire connectivity. For people who already have facilities based on Livewire IP audio, there is no need for any interface or conversion. The iPort simply plugs into a port on the Ethernet switch and all
encode/decode channels are connected with 24-bit/48kHz quality.
- Web-based configuration from anywhere on the network.
- Full range of state-of-the-art MPEG codecs. AAC-LD for delay-sensitive applications, AAC-HE and AAC-HEv2 for low bitrate requirements. Standard AAC for best quality and resilience to packet loss at higher bitrates.
- Excellent quality and high-density streaming encoder with the advantage of the Livewire IP Audio network interface.
||16 encode only
||• 1 RJ-45 - Livewire
||• 1 RJ-45 - Wide Area Network (isolated)
|Audio Conditioning (Livewire streams only)
||V-mixer: Combines streams, virtual fader. Two, 5-input mixers available.
||V-mode (8 channels available): split L/R Channels, combine L/R ... etc.
||AAC-LD for delay-sensitive applications
||AAC-HE and AAC-HEv2 for low bitrate requirements
||Standard AAC for best quality and resilience to packet loss at higher bitrates
||MPEG layer 3 (MP3) for compatibility with MP3 only codecs and software.
||2RU x 19"
||0 - 40C (32 - 104F)
||0 to 98% (non-condensing) Relative humidly
Do all the channels need to go to a single unit at the other end?
No. Each of the iPortís channels are independent and may be used individually. Simply enter the IP numbers/ports for the unit you want to use at the other end.
Can I use the iPort with Telos Z/IP or ZXS codecs at the other end?
Yes, with proper configuration.
Can I use the iPort with codecs from other manufacturers?
The iPort creates and consumes standard MPEG streams with standard RTP/UDP/IP packet formatting - nothing proprietary or special. If another vendorís codec conforms to the standards,
it should work with the iPort.
Does the iPort conform to the ITU N/ACIP specification?
No, it does not. The iPort is intended for a different class of applications. The N/ACIP standard envisions VoIP call-like operation with SIP control, whereas the the iPort is
generally used in a Ďnailed-upí way.
Our Z/IP codec family does conform to N/ACIP.
What about firewalls?
You will need to open the appropriate ports in your firewall. The IP and port numbers are easily set/determined from the iPortís Web pages, so you know which have to be opened. This
follows from the usual nailed-up applications for which iPort is intended.
Our Z/IP codec family has sophisticated technology for automatically punching through many kinds of firewalls. To do so, it uses a special server that resides outside the firewall. (This can be the one we operate as a
service to Z/IP users, or one you operate yourself.) The iPort has no way to use such a server because it does not use SIP for call set-up.
Which codec type should I use?
There are tradeoffs among those available in the iPort, with each having advantages and disadvantages. Thatís why we give you the choice. Here are some guidelines:
• AAC is the best all-round codec for bitrates of 96kbps and above (stereo). It has excellent packet loss concealment.
• AAC-HE (AAC+) should be used at rates under 96kbps. It has good audio quality at 64kbps, and is pretty good even down to 48kbps. It also has good packet loss resilience, but not as good as AAC.
• AAC-HEv2 is the most efficient codec for stereo. It has a new ďparametric stereoĒ function that kicks-in at low bitrates. Rather than sending the left/right channels discretely, it sends a core mono signal together with steering control. This makes reasonable quality stereo possible down to 32kbps, and useful stereo even to 24kbps.
• AAC-LD has the lowest delay of the codecs, so is the choice when inter-activity is important, such as for intercoms. It has about 30% less efficiency than AAC, which means that for equal quality, you would need to use 30% higher bitrate. Its packet loss concealment is good, but not as good as AAC.
• MP3 (MPEG layer 3) is not as efficient as AAC and has the worst packet loss concealment. It is included mostly for compatibility with codecs and software players that only support MP3.
Will you be including other codec types in future software releases?
Maybe. Please let us know your needs.
Where can I learn more about TCP and UDP?
Any good network engineering book would explain these in detail. One of our favorites is Computer Networking by Kurose and Ross. There is a section in our
Introduction to Livewire that introduces networking concepts to audio engineers, including a discussion of TCP and UDP. If a copy was
not included with your iPort, you can download one from our website. Indeed, the
Telos websites have a number of papers and
other resources that could be useful to you.
I need to calculate the actual network bitrate. There will be packet overhead, right?
Yes. The network rate is higher than the codec rate owing to the headers for the IP packets taking some additional bandwidth. MPEG streams are very efficient in this regard, however.
The overhead varies with the specific codec, but should be under 10%.
Will the iPort work over the public Internet?
That depends. There are no guarantees of any kind on most Internet connections. This is certainly true when multiple ISPs are involved, since nobody can take full responsibility for
the entire link. If you are lucky, all could be well. When you choose AAC as your codec, the iPort provides quite good packet loss concealment up to 10% random loss. Thatís pretty good, and would probably allow many
Internet links to work reasonably well. Higher buffer time helps, of course - but at the expense of delay.
If you can take even more delay, you can use the TCP protocol option. In this case, lost packets are recovered by re-transmission, making bad links more usable. (This, BTW, is why streaming audio over the Internet works
pretty good. The streaming servers use TCP to connect to players. Delay is not an issue - indeed, multiple seconds of buffering is the norm.)
As we mentioned before, the Telos Z/IP is intended for such applications. It has a suite of adaptive technologies to accommodate bad and variable network conditions.
What kind of data rate should I require?
As opposed to uncompressed audio that requires about 3Mbps, MPEG AAC compression typically needs about 140kbps.
About how much packet loss can be concealed?
With AAC, up to around 10% random packet loss can be effectively concealed.