Implementation of Voice over IP and Audio over IP in the Studio environment

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Mediatech 2015

Implementation of Voice over IP

and Audio over IP

in the Studio environment

Wilfried Hecht, Managing Director

AVT Audio Video Technologies GmbH Nordostpark 12

90411 Nuernberg

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Who is AVT?



The Audio product range includes

 Telephone Hybrids

 VoIP, ISDN, POTS

 High Quality Audio Codecs

 AoIP, ISDN, E1, T1

 DAB/DAB+ Headends



The first ISDN Audio Codec and ISDN

Telephone Hybrid manufacturer



First manufacturer of VoIP Telephone

Hybrids



Since 2005 member of EBU AoIP N/ACIP

standardisation group



AVT Audio Video Technologies was founded

on 1st October 1996



AVT Audio Video Technologies GmbH

manufactures high quality audio

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Telephone Hybrid Installations

HR Frankfurt

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

All dial up Audio services can be established via a PBX or outside lines



Outside lines, PBX subscriber interfaces

 POTS lines

 ISDN lines, BRI (2-B channels) or PRI (30-B channels)



Connection characteristics of all lines

 Dial up connections can be set up between subscriber units

 Synchronous connection

 Connections are traffic independent

 Low network delay

 For higher transmission data rates channel aggregation (ISDN) is needed



Audio services

 Telephony for Talk Show Systems and Telephone Hybrids with 3.1-kHz bandwidth

 Data transmission for Audio Codec communication with CD quality

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 Requirements for dial up Audio services

 Dial up Telephony connections for Talk Show applications

 Dial up connections for high quality Audio transmission

 Availability of outside lines and PBX subscriber interfaces

 Voice over IP interfaces on PBX and on DSL lines/Routers via external providers

 Internet access on DSL lines/Routers via external providers

 Voice over IP (VoIP)

 Dial up service for Telephony applications

 Standard VoIP provides an Audio Bandwidth of 3.1-kHz that POTS and ISDN provide

 HD VoIP provides an Audio Bandwidth of 7-kHz

 Asynchronous connections, Data Packets are exchanged

 Connections are influenced by the network traffic

 Variable delay determined by the network traffic (Jitter)

 Internet access

 Leased line Audio connections via IP tunnels

 Dial up Audio over IP (AoIP) connections using external AoIP providers

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Registration of a VoIP unit at a server of a VoIP provider (PBX or external)

 VoIP unit transmits its own phone number according to the SIP (Session Initiation Protocol) protocol

 VoIP provider’s server confirms the registration



Establishing a call

 VoIP unit transmits the phone number to the VoIP provider using SIP

 The VoIP provider sends an connection request to the VoIP provider of the called partner

 If the connection is possible a connection acceptance will be received

 At this moment the calling unit receives a ringing tone and the called unit rings

 The capabilities, such as coding algorithms of both partners will be exchanged using SDP (Session Description Protocol)

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Voice transmission

 The voice IP packets can be routing directly or via the VoIP provider using RTP

 RTP uses the unidirectional UDP protocol without confirmation of reception

 The advantage of UDP is a lower delay

 The delay can be influenced by the network traffic

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Voice over IP Coding algorithms

CS-ACELP: Conjugate Structure-Algebraic Code Excited Linear Prediction

AMR-WB: Adaptive Multi-Rate-Wideband MOS: Mean Opinion Score

G.711 _{(HD Voice landlines)}G.722 G.729 (AB) _{(HD Voice Mobile)}G.722.2

Algorithm PCM (A-law, µ-law) ADPCM CS-ACELP ACELP (AMR-WB)

Payload 20 msec 20 msec 20 msec 20 msec

Frequency range 300-Hz…3.4-kHz 50-Hz…7-kHz 300-Hz…3.4-kHz 50-Hz…7-kHz

Sampling frequencies 8-kHz 16-kHz 8-kHz 16-kHz

Data Rates 64-kbit/sec 48, 56, 64-kbit/sec 8-kbit/sec _{(23,85-kbit/sec)}12,65-kbit/sec

Effec. Bandwidth Ethernet ~90-kbit/sec ~90-kbit/sec ~35-kbit/sec ~45-kbit/sec

Algorithm Coding Delay 125µsec 4 msec 15 msec 25 msec

Quality (MOS) ~4.1 ~4.1 ~3.9 ~4.2

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 VoIP Audio quality is limited as HD voice to 7-kHz bandwidth  Audio over IP provides CD Audio quality for dial up connections  Defined by EBU N/ACIP group

 Extension of VoIP Standard

 SIP and SDP are extended by the Audio coding algorithms

 Fully complies with VoIP G.711 and G.722 coding algorithms (VoIP provider, PBX)

 Audio Coding algorithms

 G.711, G.722

 PCM 16/20/24

 MPEG2 Layer 2

 As Options: Enhanced apt-X, MPEG 4 Audio

 AoIP Service Providers

 PBXs do not support AoIP for external calls

 Today, external VoIP provider does not support AoIP

 Dedicated Audio over IP Service Providers are required

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Separation of “PC” network and “VoIP/AoIP” network

 Using separate network interfaces

 Using VLAN with prioritisation

 Many Switches offer an integrated Voice VLAN configuration (Cisco, HP, Netgear, etc.)

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Using QoS in the LAN

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Avoiding firewalls between PBX and VoIP systems/telephones

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Minimising the number of Switches between PBX and VoIP systems/telephones

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Do not use VPN tunnels due to higher latency and higher bandwidth requirements

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Provision of sufficient bandwidth for telephony and reservation of the bandwidth at the respective

Switch Ports

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Basically two ways of prioritisation of the “VIPs“ = “Very important packets“ are

possible

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Prioritisation on the Ethernet level (Layer 2)



VLAN (Virtual LAN)

 Each packet can get the sticker “Important” (TPID)

 Standard IEEE 802.1Q

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Prioritisation on IP level (Layer 3)



QoS (Quality of Service)

 Division into service classes

 Standard RFC3168

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Important: The prioritisation can only be guaranteed in the local network

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VLAN – Virtual network areas

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Logical sub-network within one or more

switches

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Separation of physical networks into

sub-networks

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Switches which support VLAN ensure that

packages of a VLAN are not forwarded to

another VLAN

 More efficient use of the bandwidth

 But of course no increase of the bandwidth

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4 Bytes of information are put in front of

each Ethernet packet which allows the

switch to easily allocate the packet to the

VLAN

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Signalling of a priority class with a

3 bit field

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Typical VLAN variants:

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Static VLANs – Port-based, Untagged

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Tagged VLANs

Priority Bit pattern Class of Service

0 000 No prioritisation 1 001 Background services 2 010 Reserved

3 011 General data services 4 100 Control services 5 101 Video

6 110 Voice

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Quality of Service on IP level

 Theoretically, continuous QoS end-to-end signalling is possible

 The RFC3168 Standard describes the traffic classification of the services and data streams

 Different classes for different services are possible

 Also on one network interface

 These classes are specified as Differentiated Services (DiffServ)

 In IPv4 the class is entered in the IP Header via a one Byte DiffServ field (formerly ToS=Type of Service)

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6 Bits are used for 64 different classes

(DSCP = Differentiated Services Code

Point)

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The remaining 2 Bits are used for the flow

control

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Values typically used for VoIP are:

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Voice (RTP)

 DiffServ = 184dec  Corresponds to (DSCP = 46dec) 

SIP

 DiffServ = 104dec  Corresponds to (DSCP = 26dec)

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VoIP VLAN2 PC VLAN1 Work- place

Network Concept

PABX LAN Switch untagged tagged 1 T 1 tagged 2 untagged untagged 1 QoS T tagged T QoS 2 Internet WAN LAN2 Firewall S2M/ISDN _Telecom VLAN1 2 Prio QoS 2 Prio 1 QoS 2 Prio 1 QoS 2 Prio

VLAN2: VoIP network

Layer 2: Priority class 6 (Voice) Layer 3: DiffServ RTP=184 (DSCP=46) DiffServ SIP =104 (DSCP=26) LAN1 LAN VLAN1: PC network Modem

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Tested PBXs & SIP Servers

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Hardware PBXs

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Siemens/Unify

 OpenScape Office MX V3, HiPath 3000, HiPath 8000

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Cisco

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Aastra: OpenCom

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Alcatel

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Grandstream

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