Talkswitch 24-CA - Manual (Page 187)

V O I P I N F O R M A T I O N 1 7 5
can support. Good quality VoIP is not possible over a dial-up modem
connection.
• Choose the right codec for your location
1. The default codec is G.729 (8 Kbps), using approximately 25 Kbps
bandwidth upstream and 25 Kbps bandwidth downstream for each
call. G.729 provides very good call quality while minimizing
bandwidth usage.
2. The G.726 (32 Kbps) codec is a better quality solution compared to
the G.729 codec. However, it requires more bandwidth per call. A
G.726 call typically requires 50 Kbps bandwidth upstream and
50 Kbps bandwidth downstream for each call.
3. The G.711 (64 Kbps) codec provides the best voice quality. The trade-
off is the bandwidth requirement. G.711 calls typically require up to
100 Kbps bandwidth upstream and 100 Kbps bandwidth downstream.
What happens if the power goes out or if the IP network to VoIP fails?
To ensure a reliable network connection, all elements of the VoIP network
should be connected to back-up power supplies (UPS). These elements might
include LAN switches, routers, firewalls, broadband connection devices (i.e.
cable modems, DSL modems), and VoIP devices. If the power goes out at the
Internet Service Provider, then no VoIP calls can be made. Calls can still be
placed over the regular phone lines.
Can a firewall prevent VoIP calls from passing through?
The purpose of a firewall is to control what kind of traffic enters and leaves
your network. TalkSwitch 48-CVA is designed with embedded applications to
help traverse firewalls properly. To allow VoIP calls to pass through your
firewall, you may need to use the port forwarding feature on your firewall.
TalkSwitch 48-CVA uses the following ports for VoIP:
What is SIP?
The Session Initiation Protocol (SIP) is a signalling protocol used for
establishing sessions in an IP network. A session could be a simple 2-way
telephone call or it could be a collaborative multi-media conference session.
Over the last couple of years, the Voice over IP community has adopted IP as
its protocol of choice for signalling. IP is an RFC standard (RFC 3261) from
the Internet Engineering Task Force (IETF), the body responsible for
administering and developing the mechanisms that comprise the Internet.
Format Type Unit 1 Unit 2 Unit 3 Unit 4
RTP (voice): UDP 6000-6006 6010-6016 6020-6026 6030-6036
SIP
(signaling): UDP 5060
This port is mapped to only the SIP server unit.