Voice Transmission

8/12/2019 Voice Transmission

1/20

Protocol Stack

Data Link Layer (Ethernet, ATM, FR, ...)

IP

UDP

RTP

H.26xG.7xx

Audi

o

Vide

o

RTCP

QoS-

Informations


2/20

Tasks of the various Layers

G.7xx: describes the formats for voice data(Voice- Codecs)

H.26x: describes the formats for video data

(Video-Codecs)

RTP: provides packets with a time stamp and asequence number

UDP: quick working L4-Protocol

IP: Standard L3-Protokol (Routing, etc.)

Data Link Layer: responsible for the physical

transmission and fault control, etc


3/20

H.323 Codecs

Data Voice

Network


4/20

DataIP-Header

20 Bytes

UDP-H.

8 Bytes

RTP-H.

12 Bytes

40 Bytes Overhead

Structure of a voice data packet


5/20

Interframe

Gap

Preamble

/ SFD

Destination

Address

Source

Address

Length

/ Type

Payload

Data

FCS

12 bytes 8 bytes 6 bytes 6 bytes 2 bytes 4 bytes

IP Header UDP

Header

RTP

Header

RTP

Payload

20 bytes 8 bytes 12 bytes

Overhead of a data packet


6/20

Packetising vs. Sampling

Many Samples in one Packet

good payload/overhead ratio, bandwidth savings

more delay because of longer duration for

packetising, more sensitive upon packet loss

Packetising Period Sampling time


7/20

Attributes of Voice Codecs

Codec: CodingDecoding

Bandwidth

sampling rate

quality

processing time

Standard-Codec G.711

(must be supported by all VoIP-devices!)


8/20

- 8.000 Samples/sec.- pro Sample 8 bit

-> 64 kbit/sec

Standard Voice Codec G.711


9/20

Voice Codecs

Optional (exception: G.711)

Save of Bandwidth by compressing information

Compression needs more computing power

Compression generates more delay

Compressed Signals are more sensitve upon packet loss

Terminals and Gateways must support the corresponding

compression codec (e.g. G.723, G.729A)


10/20

Pulse Code Modulation (PCM)

the analogue signal is approximated with digital values

additional noise because of the quantisation

G.711 uses PCM

no compression

provides best quality


11/20

Overview Voice CodecsCodec Algorithm Bandwidth Sampling

Time

Quality

G.711 PCM (pulse codemodulation) 64 kbit/s 0.125 ms very good

G.722 DPCM (differential pulse

code modulation)

48, 56, 64

kbit/s

0.125 ms good

G.723.1 MP-MLQ (multi pulse -

maximum likelihoodquantization) / ACELP

6.3 / 5.3

kbit/s

30 ms good- not so

good

G.726 ADPCM (adaptive

differential PCM)

16, 24, 32, 40

kbit/s

0.125 ms good not so

good

G.728 LD-CELP (low delay -

code excited linear

prediction)

16 kbit/s 0.625 ms good

G.729 CS-ACELP (conjugate

structure-algebraic code

exited linear prediction)

8 kbit/s 10 ms good

G.729A CS-ACELP (conjugate

structure-algebraic codeexited linear prediction)

8 kbit/s 15 ms statisfying


12/20

Realtime Transport Protocol (RTP)

for the transport of real-time data (Audio, Video) over

paket-oriented network

NO QoS- guarantees

Unicast and Multicast

supplement by RTCP

independent from L4/L3 (normally over UDP/IP)


13/20

Features of RTP

Sequence number to guarantee the right sequence on

the receiver side

Timestamp for keeping the original time-lags in

the data packet sequence

No Reassembling

No error correction


14/20

RTP Header

V P X CC M PT Sequence Number

Timestamp

Synchronization Source Identifier (SSRC)

Contributing Source Identifiers (CSRCs), optional

V: Version number present state: Version 2 (2 bits)

P: Padding Bit indicates fill bytes in the payload (1 bit)

X: Extension Bit indicates an opitonal header extension (1 bit)

CC: CSRC Counter Number of Contributing Sources (4 bits)

M: Marker Interpretation depends on the PT (1 bit)

PT: Payload Type (7 bits)

0 2 3 4 8 9 16 32

fixed

header

extension


15/20

Realtime Control Protocol (RTCP)

Control Protocol for RTP

UDP: RTP even Port Number (x)

RTCP odd Port Number (x+1)


16/20

Tasks of RTCP

provides informationen about the current transmission quality

(Sender Reports SR; Receiver Reports RR)

identifies the RTP-Source (Canonical Name CNAME)

selfcontrol (BandwidthRTCP 5% BandwidthRTP)

supplementary infromation about the subscriber (optional)


17/20

RTCP Messages

SR - Sender Reportcontains statistics of the receiver and the

sender

RR - Receiver Report

contains receiver statistics of subscribers

that are not sending

SDES - Source Description

contains the CNAME (canonical name)

BYEindicates the end of a participation

APPapplication specific functions


18/20

RTCP Header

V P Count Length

Report Block

V (Version) present state: Version 2RTP (2 bit)

P (Padding) indicates fill bytes in the payload (1 bit)

Count number of the received report blocks (5 bit)

Type RTCP Paket-Type, for instance SR or RRLength length of the RTCP-packets (in multiples of 32 bit)

Sender Info Block for Sender Report only

0 2 3 7 15 32

Type

Sender Info Block (optional)


19/20

Sender Info Block

Sender Report contains Sender Info Block

if at last a RTP-Paket was sent from the sender

(no SR or RR)

contains statistical informationen about the RTPpackets that have been sent out, e.g.

- number of RTP packets sent

- number of payload octets sent.


20/20

Report Block

forwarded to receiver from sender

contained in every RTCP-Paket

contains statistical informationen about receivedRTP Packets, e.g.

- overall number of lost RTP packets since last report

- jitter, etc.

Documents

Voice Transmission