
June 2015 - Ed.2.1
Audio over IP decoders E411, E413, E415 & E417 technical manual
Page 3
Chapter 2: Functionality
2.1
Audio source types
Decoders can play high quality stereo sound coming from several source types:
RTP:
Audio streams coming from E321 encoders, E495 or E497 paging consoles, E430 audio server, computer
applications or other devices, in the RTP format as defined in RFC 3550.
The RTP can be received also in multicast mode. So, many decoders can play the same RTP stream
without extra bandwidth consumption.
The allowable payload formats are:
o
G711-A: (vocal quality, VoIP standard)
o
PCM 16 bits/sample, 1 or 2 channels, 11025 Hz (voice/music – medium quality)
o
PCM 16 bits/sample, 1 channel, 22050 Hz (voice/music – good quality)
o
PCM 16 bits/sample, 1 channel, 32000 Hz (voice/music – very good quality)
o
MP3 (quality according to emitter)
When using these RTP formats, as audio is not compressed, the latency is fairly low
1
.
RTP/SIP
2
:
Audio streams coming from a standard SIP phone or PBX. The audio codec used is G711-A.
The decoder has an internal SIP agent that can be registered in an SIP PBX and acts as a standard
extension accepting calls.
Also, can be configured in “peer-to-peer” (P2P) mode if not PBX is used to accept calls directly from
phones.
When a call is received, is automatically answered and the audio is sent to the output if no other higher
priority channel is being played.
SHOUTcast / ICEcast
3
:
High quality audio streams coming from local or internet servers (internet radios). These streams can
also be generated with the ENA software or the E430 audio server.
The device connects to the configured URL to acquire the compressed audio stream. The audio can be
sent in any of the following formats:
o
MP3 up to 192 kb/s
o
AAC+ up to 192 kb/s
o
Ogg-Vorbis up to 192 kb/s
Due to audio compression and buffering, this format has a very high latency, that can reach several
seconds.
1
The minimum latency is obtained in a well dimensioned and configured network, and can be as low as 40 milliseconds.
2
SIP is the most widely used protocol nowadays for session establishing in VoIP telephony. It is defined in RFC 3261 .
3
SHOUTcast and ICEcast are both standards used to transmit music through Internet. There are many public radios in Internet
using these protocols, and is quite easy to create a proprietary radio using one ICEcast server provided by an ISE (Internet
Services Provider).