Book Image

Building Telephony Systems with OpenSER

Book Image

Building Telephony Systems with OpenSER

Overview of this book

Table of Contents (18 chapters)
Building Telephony Systems with OpenSER
Credits
About the Author
About the Reviewers
Preface
6
Building the User Portal with SerMyAdmin
Index

Understanding SIP and RTP


To understand the following subsections, you should understand some things about SIP and RTP. First, SIP is a signaling protocol that controls the call with methods such as INVITE, BYE, and CANCEL. The SIP protocol includes in the INVITE request information about the session (audio/video/text) using a protocol called SDP (Session Description Protocol). The information contained in the SDP describes one or more media flows configured between two user agents.

A SIP proxy never participates in the media flow, thus it is media agnostic. In other words it supports calls with whichever media are specified by UA and gateways. However, sometimes a B2BUA (back to back user agent) such as mediaproxy can be installed at the same server to treat RTP audio (that is, NAT traversal mechanism). The SDP protocol works in an Offer/Answer model. The SDP offer is embedded in the INVITE request and the answer in the 200 OK response.

Example: Excerpt from Ethereal:

The packet described...