This is the concluding part of the article "Voice over IP Basics for IT Technicians - Part One". Read the article "Voice over IP Basics for IT Technicians - Part One" before you start reading this one.
VoIP networks
Local (LAN) and wide area networks (WAN) can both support VoIP operation. However, there are significant differences between LAN
and WAN performance that affect signaling execution speed and voice quality. The LAN implementation uses Ethernet as the transport
network. VoIP devices operate following the IEEE 802.3 standards. No proprietary changes have been made to the Ethernet protocols
and architecture. The Ethernet LAN operates at 10 and 100 Mbps on networks with very short delay, no jitter (delay variance), few
errors and no packet loss. Although VoIP traffic could share the LAN with data users, it is recommended that the voice and data
devices run on separate VLANs (IEEE 802.1q) on the LAN switches for both performance and security reasons. Voice quality and signaling
execution speeds are as good as the TDM PBXs.
The WAN presents a number of performance-hindering issues. Bandwidth is limited, end-to-end delays are much longer, packet jitter
occurs and, finally, packets are lost. The transmission error rate is low enough that signaling packets do not often require retransmission,
and voice quality is not impaired by errors. The end-to-end delay goal between IP phones is 150ms. The receiving IP phone must
compensate for jitter by waiting for all the packets of a word to arrive before the word can be converted back into analog sound. The
receiving IP phone must also insert simulated voice packets to eliminate holes in a word, which occurs when packets are lost.
Correcting jitter and packet loss causes extra delay between IP phones.
Extra bandwidth and Quality of Service (QoS) techniques can solve these problems. Voice calls consume bandwidth: about 80 Kbps
when no voice compression is used (G.711) and about 25 Kbps (G.729) when the voice is compressed. The actual bandwidth consumption
will vary based on compression type and packet size. Bandwidth is probably not an issue on the LAN as most LANs operate at low
utilization, probably less than 10% to 20% utilization. If bandwidth is not increased on the WAN, then voice and data users will suffer
call degradation.
QoS can be supported on LAN switches using the IEEE 802.1p standard. This standard requires that IP phones and gateways also support
the 802.1p standard. Routers produce QoS through the implementation of DiffServ, which must be supported by the IP phones and
gateways, as well as MPLS in the routers. RSVP was an early technique for VoIP QoS, but is less frequently used in todays products.
How VoIP works
A VoIP-based PBX starts up (boots up) like other servers. Once the booting up is complete, the IP phones and gateways can register
with the call server (see Figure 1 in Part One). The IP phone and/or gateway must first access a DHCP to obtain an IP address. The DHCP may be
part of the data network, or it may be a separate server, or it can be integrated with the call server. Once an address has been
assigned, the IP device contacts the call server to register. The call server may have a common set of privileges and restrictions for IP
devices or an administrator can make the feature assignments.
The call server or another assigned server also adds this device and its phone number(s) to the DNS to support directory services. A
permanent H.323 TCP session is established between the VoIP device and the call server. This is true for most proprietary signaling
protocols. SIP uses UDP for the signaling path.
When a user picks up the phone, the dial tone can be generated locally by the phone or by the call server. The IP device then sends
one or more packets requesting a connection and the features to be implemented during the connection, such as a conference call.
The call server then determines whether the other device is available or busy. If available, the call server contacts the receiving device
and instructs both the caller and called devices to establish a peer-to-peer UDP path to carry the RTP speech. The call server becomes
dormant during the call until one of the devices terminates the call. The call server then breaks the peer-to-peer connection and
records the call event as part of the Call Detail Record (CDR).
Security of VoIP has become a major concern for VoIP adopters. Most firewalls do not support VoIP except through VPN connections.
As a default, data firewalls will probably prevent the operation of VoIP by users on the untrusted network when they call devices on
the trusted network. Several vendors are now offering encrypted signaling and encrypted speech behind the firewall. This design prevents
security problems produced by users of the trusted network. The encryption functions must be implemented in the IP phones,
gateways and call servers.
Conclusions
VoIP forces an expanded role for the network technician. Not only will the physical and LAN deployment and troubleshooting tasks
continue to exist, but there will be new responsibilities. These will include VoIP protocol operation, call server interaction, network
performance measurement and IP phone configuration. Compounding these responsibilities are the vendors who have chosen to implement
multiple standards as well introduce proprietary solutions to the VoIP mix. The job description for the future network technician
will have to expand to meet these new challenges.
