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Standard Algorithm Transmission Rate MOS

The document describes several audio coding standards used for voice transmission over IP networks: - PCM (G.711) provides basic uncompressed coding at 64 kbps with good quality but high bandwidth. ADPCM (G.726) reduces the bit rate to 32 kbps with slightly lower quality. - CS-ACELP (G.729) and ACELP (G.723.1) use more advanced compression to achieve even lower bit rates of 8 kbps and 5.3 kbps respectively, with quality scores slightly below PCM/ADPCM. - RTP is used as the transport protocol to ensure on-time delivery, sequencing and error correction of voice packets transmitted over

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49 views3 pages

Standard Algorithm Transmission Rate MOS

The document describes several audio coding standards used for voice transmission over IP networks: - PCM (G.711) provides basic uncompressed coding at 64 kbps with good quality but high bandwidth. ADPCM (G.726) reduces the bit rate to 32 kbps with slightly lower quality. - CS-ACELP (G.729) and ACELP (G.723.1) use more advanced compression to achieve even lower bit rates of 8 kbps and 5.3 kbps respectively, with quality scores slightly below PCM/ADPCM. - RTP is used as the transport protocol to ensure on-time delivery, sequencing and error correction of voice packets transmitted over

Uploaded by

karamdo
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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Standard

G.711
G.726
G.729
G723.1

Algorithm
PCM
ADPCM
CS-ACELP
ACELP

Transmission Rate
64
32
8
5.3

MOS
4.1
3.85
3.92
3.56

----------------------=============================================
constant bandwidth, guaranteed time of packet delivery (also called jitter) and
correct sequence are necessary for successful voice transmission. We need not
worry about delivery of any packet during voice transmission because mathematical
methods used for voice signal coding and decoding can make approximation when a
packet has not been delivered. Thus, we can use the UDP for voice stream
transmission, which has no acknowledgement of delivered packets, but in any case
we need a protocol that is responsible for voice coding, jitter, sequence order and
bandwidth. This protocol is called RTP (Realtime Transport Protocol) and is widely
used for voice transmission in modern VoIP networks.

2.2 SIP as a Signalling Protocol


The SIP (Session Initiation Protocol) is a text-based protocol, similar to the HTTP
and SMTP, designed for initiating, maintaining and terminating of interactive
communication sessions between users. Such sessions include voice, video, chat,
interactive games, and virtual reality.
The SIP defines and uses the following components:

UAC (User agent client) client in the terminal that initiates SIP signalling

UAS (User agent server) server in the terminal that responds to the SIP
signalling from the UAC

UA (User Agent) SIP network terminal (SIP telephones, or gateway to other


networks), contains UAC and UAS

Proxy server receives connection requests from the UA and transfers them
to another proxy server if the particular station is not in its administration

Redirect server receives connection requests and sends them back to the
requester including destination data instead of sending them to the calling
party

Location Server receives registration requests from the UA and updates the
terminal database with them.

All server sections (Proxy, Redirect, Location) are typically available on a single
physical machine called proxy server, which is responsible for client database
maintenance, connection establishing, maintenance and termination, and call
directing.
Basic messages sent in the SIP environment

INVITE connection establishing request

ACK acknowledgement of INVITE by the final message receiver

BYE connection termination

CANCEL termination of non-established connection

REGISTER UA registration in SIP proxy

OPTIONS inquiry of server options

Answers to SIP messages are in the digital format like in the http protocol. Here are
the most important ones:

1XX information messages (100 trying, 180 ringing, 183 progress)

2XX successful request completion (200 OK)

3XX call forwarding, the inquiry should be directed elsewhere (302


temporarily moved, 305 use proxy)

4XX error (403 forbidden)

5XX server error (500 Server Internal Error, 501 not implemented)

6XX global failure (606 Not Acceptable)

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