13.6.4  
Voice over IP (VoIP) 
 
Internet telephony features are described in detail in the feature description.  
Voice over IP (VoIP) enables voice data to be transmitted over an IP network, both internally and via the Internet.
Voice over IP (VoIP) is an outstanding feature of HG 1500. In combination with the clients connected to the LAN, the board makes it possible to use not only VoIP but also the telephony features of the system. In particular, this feature can be used for teleworking. This means that full Voice over IP functionality is available at the teleworking PC.
As of the HiPath 3000 V3.0 software version in combination with new boards (HXGS3, HXGM3, HXGR3), login is no longer performed under "Voice Gateway" in HiPath 3000 Manager I but in HiPath 3000 Manager E.
The following combinations of VoIP connections via the HG 1500 are possible:
 •   IP to IP, for example, subscriber C to subscriber D or subscriber D to subscriber F.
 •   IP to local SCN, for example, subscriber D to subscriber B.
 •   Local SCN to IP, for example, subscriber B to subscriber C.
 •   IP to PSTN, for example, subscriber C to subscriber E.
 •   PSTN to IP, for example, subscriber E to subscriber D.
Interworking is fully supported.
With the exception of IP to IP connections, one DSP channel per connection is required for each HG 1500 involved. Conferencing requires one DSP channel for each SCN subscriber.
In IP networking, SCN to SCN connections (local or PST) also can be run transparently via the HG 1500. This is described in Section 13.6.3 "IP Networking".  
This can lead to significant call charge savings if an appropriate IP network is available. The user can also avail of telephony features, such as name transmission and callback, in the usual way.
The hardware prerequisite for IP networking is that the systems involved must have at least one of the following HG 1500 boards:
 •   STMI2: Up to eight boards can be used per system (max. four boards per HiPath 3800 cabinet).
 •   HXGS (wall housing) or HXGR (19" housing): Up to four boards can be installed in HiPath 3500/3550 and up to two in HiPath 3300/3350. Both boards can process up to 16 IP channels for voice connections with a single PDM1 expansion module.
The IP networked systems and the HG 1500 boards are administered with two different tools:
 •   HiPath 3000 Manager E: Extending the existing HiPath 3000 customer database to include information for IP networking.
 •   HG 1500 WBM: Performing HG 1500 board settings.
The IP address used for IP networking is always that of the first LAN interface. If IP networking is routed via another interface, such as LAN2, use IP routing entries to ensure that the LAN1 interface can be reached.  

13.6.4.1  
General Parameters for Voice over IP 
Before configuring system clients, you must configure the valid parameters for both variants in HiPath 3000 Manager E:
 •   Echo:
You can define here how to compensate for the echoes that occur in connections to conventional telephones via the voice gateway.
 •   Traffic statistics:
This is where you define whether to record traffic statistics which can then be queried via SNMP.
 •   Encoding:
This parameter is used to select which voice coding will be implemented.

13.6.4.2  
Voice Coding 
HiPath 3000/5000 supports voice compression according to the G.729 and G723.1 standards. G.723.1 facilitates the use of teleworking via ISDN for example, and reduces the load on the existing data network. G.723.1 reduces the VoIP bandwidth requirement for voice communication. This enables a bidirectional voice connection via an ISDN B channel. This, in turn, facilitates the connection of teleworkstations or branch offices to the company network via a 64 KB connection. However, it is not advisable to operate other applications (such as Application Sharing, File Transfer) in addition to a bidirectional VoIP connection simultaneously via 64 Kbps since this generally leads to a deterioration in voice quality. It is recommended that both B channels of an S0 connection are used for this purpose.
The HG 1500 supports codecs for compression and decompression of voice signals in accordance with industry standards G.711 A-law, G.711 µ-law, G.723, G.729A and G.729AB. The codecs can be arranged according to their priority. The gateway automatically attempts to use a codec with the highest possible priority for each connection.
The HG 1500 supports the following codecs:
 •   G.711 (A-law and µ-law) voice coding - an ITU (International Telecommunication Union) standard for voice coding at 56 or 64 Kbps
 •   G.723.1 voice compression - an ITU standard for voice coding at 5.3 or 6.4 Kbps
 •   G.729A and G.729AB voice compression - an ITU standard for voice coding at 8 Kbps

13.6.4.3  
DTMF Handling 
DTMF (Dual-Tone Multifrequency) is the type of audio signal used for tone dialing in accordance with ITU-T standard Q.24. It can be used for call setup and for signaling during a call.
HG 1500 supports the detection and processing of DTMF (Dual Tone Multi Frequency) signals in accordance with Q.24 regarding signaling for VoIP calls. In particular, it supports the following functions:
 •   Detection of DTMF signals
 •   Conversion of DTMF signals into user input signals in accordance with H.245
 •   Generation of DTMF signals for PSTN
 •   Speech protection using Bellcore test tapes

13.6.4.4  
Voice Quality 
Various HG 1500 functions ensure optimum voice quality during VoIP connections. Further details are described below.

Quality of Service (QoS)  
QoS enables voice packets to be given higher priority than other packets in the IP network so that voice quality is influenced as little as possible by other data traffic.
To ensure the best possible quality especially for voice connections, the HG 1500 supports various QoS protocols:
 •   IEEE802.1p Tags (level 2)
 •   Type of Service (ToS) prioritization (RFC 791, layer 3)
 •   Differentiated Services (DiffServ; RFC 2474, layer 3)
The gateway observes these standards for routed packets and makes them available for applications on the gateway.

Echo suppression  
The HG 1500 supports echo suppression in accordance with G.168 to ensure acceptable voice quality in the WAN. Echo suppression is effective for round trip delays of up to 48 msec.

Dynamic jitter buffer  
The dynamic jitter buffer eliminates the distortion caused by fluctuations in the arrival of voice and video packets. Individual packet delays can be a result of network overload or internal processing within the LAN endpoints.
The HG 1500 jitter buffer can be configured for the connection requirements of a specific network.

Voice Activity Detection  
VAD (Voice Activity Detection) checks all incoming signals for voice signals. If a signal exceeds a preset level, the VAD recognizes it as voice and anything under that level as silence.
The HG 1500 provides VAD and silence suppression for all supported codecs. When no voice signal is detected, a background noise is inserted at each end of the line to emulate traditional telephone line conditions.

13.6.4.5  
Environmental Requirements for VoIP 
To ensure voice transfer quality and avoid unacceptable delays, the networks being used must meet certain requirements.

Environmental Requirements in the LAN  
LANs used for VoIP must meet the following specifications:
 •   At least 256 Kbps transmission capacity per networked unit
 •   Not more than 50 msec delay in one direction (One Way Delay) Not more than 150 msec total delay
 •   Not more than 1% packet loss
 •   QoS support - IEEE802.1p, DiffServ (RFC 2474) or ToS (RFC 791)
 •   Every HG 1500 must be connected via a switch or a dedicated port of a router.
 •   It is recommended that the VoIP application be connected via a separate VLAN to reduce collisions with other transmissions. If all involved devices support VLAN (in accordance with IEEE802.1q), all VoIP traffic can be placed in a separate VLAN. For administration access in this case, LAN switches must provide individual computers with access to multiple VLAN segments.
 •   Not more than 20% of the available bandwidth should be used.
 •   Not more than 10% of the total data traffic should be broadcast packets.
 •   The error rate should not amount to more than 1% of data traffic and should not tend to increase.

Environmental Requirements in the WAN  
LANs that are linked over WANs and share the same VoIP functionality must meet the following minimum requirements:
 •   Each LAN must each be connected to the Internet via xDSL with a fixed IP address.
 •   QoS support - IEEE802.1p, DiffServ (RFC 2474) or ToS (RFC 791) - over the entire connection
 •   The bandwidth required for the calls must always be available in both directions, to the network and to the user.
 •   Not more than 50 msec delay in one direction (One Way Delay) Not more than 150 msec total delay
 •   Not more than 3% packet loss
 •   Not more than 3% error rate
 •   Not more than 10% jitter
 •   As little broadcast and multicast traffic in the network as possible. If necessary this can be achieved by structuring the network - using VPN, for instance - with Layer 3 switches and routers, or with Layer 2 switches that recognize multicasting.
 •   Not more than 40% network load (without VoIP traffic)
 •   Less than 40 broadcast packets per second if possible

Environmental requirements for fax/modem over G.711  
Fax/modem over G.711 can be enabled if the IP network satisfies the following requirements:

 Table 13-3   Prerequisites for Modem over G.711
Value  
Time  
Jitter  
<= 20 msec  
Delay (one way)  
<= 50 msec  
Delay (round trip)  
<= 100 msec  
Packet drop  
<= 0.05%  
If the above values are not observed in the IP network, the fax or modem transmission may be interrupted.

HiPath 3000/5000 V9, Service Documentation, Issue 10   up Top
Disclaimer & Copyright   ID: P31003H3590S100017620   2014-02-27