IP Audio Compact
UDP, RTP, SIP… – IP Protocols, what are they all for?
The chosen protocol determines mainly the characteristics of the audio transmission in a network. Due to the fact that there are high requirements regarding the signal quality, the appropriate protocol needs to be selected. RTP/UDP is typical for Corporate Networks and VPNs with dedicated communication partners, while SIP is the protocol used on the public Internet; it is particularly popular for VoIP applications.
SIP Session Initiation Protocol
Session Initiation Protocol (SIP) is a text-based protocol, for negotiation of connections based on the Internet Protocol (IP). SIP is used merely to handle the signalling between individual negotiation parties. The transport of media data runs – similarly to ISDN – separately from the negotiation. The media data are often sent over a different route, conveyed by a different Transport Protocol. If TCP or UDP are available for the negotiation, RTP is mostly used for the media data transport.
SIP-connections as substitution for ISDN
The new SIP connections in the public Internet can be considered as equivalent to ISDN if there is a loss free transmission with low latency. This can be achieved with guaranteed bandwidth, e.g. using RSVP and FEC.
VPNs with guaranteed bandwidth as substitution for E1 and X.21 connections
Virtual Private Networks, so called VPNs can substitute todays E1 or X.21 connections, if a stable and guaranteed bandwidth is achieved. Using IP in existing E1 or ATM networks.
The IP-protocol can also be applied in existing E1 or ATM connections
This allows the simultaneous use of audio/video and data signals. The reservation of the corresponding bandwidth has to be done within the routers.
HE AACv2 etc - Audio Coding Formats, which one for which application?
The huge number of audio coding formats can be confusing sometimes. The selection needs to be defined based on criteria like compatibility , quality and latency. While MPEG 4 HE AACv2 is providing excellent quality at very low bit rates at e.g. 48kbps stereo, transparent AES/EBU transmission is used in networks with more bandwidth to be completely loss less and obtaining production quality with up to 3 MBit/s.
Migration from ISDN to IP and vice versa
For the migration process of existing ISDN to IP infrastructure, a certain time will pass. Therefore technical solutions are required which allow transcoding of formats and bridging of networks, so called Gateways.
FEC – Forward Error Correction
FEC permits error detection and/or correction by adding redundant data to the stream. Thereby avoiding re-transmission or corruption of data and the associated costs of higher bandwidth needs and increased delay.
IP Overhead
At IP-transmissions the data stream consists of Payload (pure audio data e.g. one or more MPEG frame) and IP Overhead. In case of UDP protocol the absolute overhead is as large as 46 Bytes / packet and it consists of 8 Bytes UDP header, 20 Bytes IP header and 18 Bytes IEEE802.3 (Ethernet). RTP protocol has 58 Bytes overhead per packet. 12 Bytes - RTP, 8 Bytes UDP, 20 Bytes IP and 18 Bytes IEEE802.3. The relative IP Overhead is calculated in percents of the original payload.
MPEG TS via ASI, MPEG TS via IP
The MPEG Transport stream, named MPEG TS is used mainly in DVB and ipTV environments. It is a format containing one or more audio and/or audio/video streams which, typically, need to reach set-top boxes as final receivers.
Are standards available?
In order to result in common implementations, the EBU has established the working group N/ACIP. In September 2007 the first final recommendation shall be officially released.
RTP (Realtime Transport Protocol)
is a session protocol for IP-based networks which uses UDP (User Datagram Protocol) as transport protocol. In opposite to the pure UDP implementation RTP guarantees the right sequence of packets at the receiver side.
RTCP (RTP Control Protocol)
is working on top of the RTP. Not interfering with a real data stream it provides some control data exchange between streaming session participants. Receiving a feedback information as packets sent, lost packets, round trip delay, jitter, etc., the streaming can be accommodated on current network conditions.
SDP (Session Description Protocol)
Descriptions of multimedia streaming sessions with SDP include important initialization parameters information. They are being used to invite participants and to announce the start of the session or to initialize the connection on some other way.
SAP (Session Announcement Protocol)
is used to announce the information provided by SDP on some well known multicast address to give the potential clients an overview of the contents available on the current (sub) network. MAYAH supports in addition a non traditional way for SAP unicast.
RTSP (Real Time Streaming Protocol)
is an application protocol for use in streaming media systems. It lets client to know about the contents available for streaming and to choose the appropriate stream as well as to control a streaming media server with a typical commands such as “play” or “pause”.
