Frequently Asked Questions - Voice Over IP
||What does the VoIP require for the local and remote connection?
||VoIP is supported on the CallHandler using dedicated VoIP cards. Each card
can carry 60 simultaneous full-duplex voice channels, and up to four cards
can be supported at present giving 240 VoIP channels per PC chassis.
To connect to a VoIP channel on the CallHandler you require either an
IP phone or a multimedia PC running SoftPhone software.
This is the list of compatible (tested) client software:
- Netmeeting (Low quality)
- OpenH323 (We have this running)
This is the list of compatible (tested) phones:
- Cisco (But you need the Cisco Call Manager)
- PhoneJack (This is a card you plug in your PC, then plug a phone into the card)
|Is it portable? That is, could a laptop be set up with the capability and a connection be made via dialup to the net?|
| ||Yes, this is possible, but the voice quality is likely to be low. This
is because a modem link is slow (slow data links = poorer voice quality),
and the Internet data speed is very intermittent, it is very dependant
on other network traffic.|
|Is it Scalable and what are the limitations?|
| ||At the moment we can fit 240 VoIP channels in a chassis, and we can cluster
systems together using a TCP/IP messaging system for control, and a fibre
optic link (MC3) between systems for transferring calls between chassic to create a very large system.|
|How is the quality of the connection? I imagine it is good in an
| ||This is a big subject:
The quality is dependent on three things:
- Bandwidth - Do you have enough network bandwidth to carry the number of
voice channels you need?
- Codecs - Choice of which codecs you allow to connect: G711 Best Quality
equivalent to ISDN->ISDN call, G729A/B Good Quality,
G723.1 Medium to Poor Quality.
- Network - Mainly other network traffic, which introduces Delay, Jitter,
and at worst Lost Packets.
On internal networks, where traffic is planned and managed you can get really excellent quality, quite often better than existing analog extensions.
Over the internet, can get reasonably good results, depending on other traffic, time of day etc., and providing you have a good broadband connection.
Using dial-up direct to a CallHandler server, can get fairly good results (equivalent to a mobile phone) provided there is no other network traffic.
Using dial-up over the Internet, usually poor, the best that is possible is a Medium quality connection.
|Does your system provide call progress analysis? That is, can it count the number of rings for a no answer? detect operator intercepts (special information tones), answering machines, busy and of course voice level energy?|
| ||Ring Timeout: We actually time the call, this can be done to the nearest millisecond.
Network Codes: We trap all call progress messages sent by the network, the response to these messages can be configured.
Answering Machines: We haven't enabled answer machine detection because it takes time to detect the answering machine 3-5 secs, on all calls. Our customers so far prefer for the agent to catch the first "Hello"
Busy: Yes, we can also distinguish between Network/Fast Busy and Normal Busy signals.
Voice Energy: We have the facility to detect voice energy, but we don't use it.
|Can you give me more information on video over Voice Over IP (VoIP), this is a real benefit in
the call centre or strategic customer service environment. i.e. picture and
voice point to point.|
Both the H323 and SIP standards support the streaming of video.
H323 and SIP are basically both protocols for establishing streaming
connections between 2 (or more parties). Both H323 and SIP use the
RTP (Realtime Transport Protocol) and RTCP (Realtime Transport Control
Protocol) for streaming. RTP can stream just about any format of audio
and video. So the protocol is pretty much in-place to support any sort
of multimedia. Comparing audio and video streaming, a typically compressed
audio stream uses 8kbits/sec, whereas a typically compressed video stream
uses typically 64-128kbits/sec. Compressing and streaming video around
and enterprise is a much larger problem than audio, you need much bigger
network bandwidth, and much bigger codec cards, and hence is a lot more
What about security and Voice Over IP (VoIP), will or could the connectivity of telephone
networks to IP present some sort of security breach to the corporate
| ||There are several issues with security:
i) At present with most VOIP systems it is not possible to encrypt the
audio channels, so anyone with a VOIP network monitor can listen in on
a VOIP conversation.
ii) H323 is not very good at connecting through firewalls. The H323
handshake is done via one IP port, then the actual stream connection is
made on another port, which is agreed upon dynamically via the handshake
process. To make connections through a firewall it is essential to know
which port connections to allow and which to deny, and H323's dynamic
port allocation makes this pretty much impossible. There are a couple of
products that get around this problem, but they're not very mature yet.
This is not an issue with SIP as the handshake and stream connections
are made on fixed pre-defined ports.
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