VoIP White Paper
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© 2006 InSciTek Microsystems, Inc. All rights reserved. Allworx is a registered trademark of InSciTek Microsystems. All other names may be
trademarks or registered trademarks of their respective owners.
Revised: February 8, 2007
Page 4
4 SIP Protocol and VoIP
The Allworx VoIP platform is built around the industry standard VoIP protocol known as Session Initiation
Protocol (SIP). SIP is a packet-based protocol built on top of the standard IP stack using the User Datagram
Protocol (UDP/IP). Although it is possible for VoIP telephony to use other standards (e.g., H.323 or MGCP),
SIP was specifically designed for IP stacks, and was developed with Internet protocols in mind. While
historically much of the older installed base of packet-based telephony used MGCP or H.323, it is generally
accepted that SIP represents the future of VoIP – and nearly all-new installations using industry-standard
protocols are deployed using SIP.
When it comes to IP-based VoIP, SIP is not the whole story – it actually describes only one of the three
functional elements required. In other words, when designing a VoIP protocol, three basic functions must be
provided:
1.
Call Control and Setup/Termination – A set of mechanisms to locate the intended dialed parties,
determine their availability, and accept or deny their requests.
2.
Session Negotiation – Once a new call is going to be accepted, this determines the format and network
locations for transporting audio between the actual end points.
3.
Media Transport – Once accepted and negotiated, this provides real-time audio transport between the
end-points for the call ‘s duration.
SIP itself actually only provides the first item described above. Other protocols are actually required to perform
the other two primary functions. When people talk about SIP, two protocols are also generally implied:
• Session Description Protocol (SDP) to negotiate the session media types (G.711 or G.729) and the IP
address and port number that each end-point should transmit toward.
• Real-Time Transport Protocol (RTP) to actually move coded audio data during the live call.
Therefore, when people talk about SIP VoIP telephony, several things are implied to be available and working
properly for successful phone calls:
• Reliable IP routing data connectivity of UDP packets between associated phones and their gateways or
proxy servers for basic network transactions (network settings, DHCP, DNS, etc.).
• SIP protocol and proxy configuration to locate intended parties and determine their availability (ringing or
busy).
• SDP negotiation to determine the final coder type, IP addresses, and port numbers that should
communicate actual audio data.
• RTP to transport coded audio over a network with an acceptable QoS level from end-to-end.
Acting as a VoIP gateway, the Allworx server contains all the necessary facilities to make the above happen in
as simple a manner as possible. However, when one of these mechanisms is interfered with on the network,
certain types of symptoms may result – such as dropped calls, choppy audio, one-way audio, or echo.
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