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

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1 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 Nuernberg Germany Web:

2 AVT Audio Video Technologies was founded on 1st October 1996 AVT Audio Video Technologies GmbH manufactures high quality audio transmission products 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 2

3 WDR Cologne SWR Stuttgart HR Frankfurt Telephone Hybrid Installations 3

4 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 Present Broadcast Stations 4

5 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 Broadcast Stations in the future 5

6 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) 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 Voice over IP Dial up Procedure 6

7 G.711 G.722 (HD Voice landlines) G.729 (AB) G (HD Voice Mobile) 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 12,65-kbit/sec (23,85-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 Costs/Patents Free Free USD Initial ca per channel Yes CS-ACELP: AMR-WB: MOS: Conjugate Structure-Algebraic Code Excited Linear Prediction Adaptive Multi-Rate-Wideband Mean Opinion Score Voice over IP Coding algorithms 7

8 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 Audio Over IP Standard 8

9 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.) Using QoS in the LAN Avoiding firewalls between PBX and VoIP systems/telephones Minimising the number of Switches between PBX and VoIP systems/telephones Do not use VPN tunnels due to higher latency and higher bandwidth requirements Provision of sufficient bandwidth for telephony and reservation of the bandwidth at the respective Switch Ports LAN Network Requirements 9

10 Basically two ways of prioritisation of the VIPs = Very important packets are possible Prioritisation on the Ethernet level (Layer 2) VLAN (Virtual LAN) Each packet can get the sticker Important (TPID) Standard IEEE 802.1Q Prioritisation on IP level (Layer 3) QoS (Quality of Service) Division into service classes Standard RFC3168 Important: The prioritisation can only be guaranteed in the local network Prioritising of VoIP packets 10

11 Logical sub-network within one or more switches Separation of physical networks into subnetworks 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 4 Bytes of information are put in front of each Ethernet packet which allows the switch to easily allocate the packet to the VLAN Signalling of a priority class with a 3 bit field Priority Bit pattern Class of Service No prioritisation Background services Reserved General data services Control services Video Voice Network control Typical VLAN variants: Static VLANs Port-based, Untagged Tagged VLANs VLAN Virtual network areas 11

12 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) 6 Bits are used for 64 different classes (DSCP = Differentiated Services Code Point) The remaining 2 Bits are used for the flow control Values typically used for VoIP are: Voice (RTP) DiffServ = 184dec Corresponds to (DSCP = 46dec) SIP DiffServ = 104dec Corresponds to (DSCP = 26dec) Quality of Service on IP level 12

13 PC VLAN1 VoIP VLAN2 VLAN VLAN1: PC network untagged Workplace tagged 2 Prio QoS 1 2 Prio QoS LAN tagged 2 Prio QoS VLAN2: VoIP network Layer 2: Priority class 6 (Voice) Layer 3: DiffServ RTP=184 (DSCP=46) DiffServ SIP =104 (DSCP=26) 1 T 1 2 LAN Switch T T Firewall Modem Internet untagged LAN2 LAN1 untagged 1 tagged WAN QoS 2 Prio QoS PABX S2M/ISDN Telecom Network Concept 13

14 Hardware PBXs Siemens/Unify OpenScape Office MX V3, HiPath 3000, HiPath 8000 Cisco Aastra: OpenCom Alcatel Grandstream Software SIP Servers Brekeke (VoIP & AoIP) 3CX (VoIP) Asterisk (VoIP) Conclusion Almost all PBXs/SIP Servers show different behaviour even if only in some aspects Interworking tests are required Tested PBXs & SIP Servers 14

15 Thank You! See our VoIP and AoIP products live at stand H17/4 Further information: 15

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