Configuring ScreenOS Devices Guide
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Configuring SIP Settings
116
Click Add. The New Pre-defined Service Session cache dialog box appears.
5.
NOTE: All predefined services will be displayed in the dialog box. Click
Edit Pre-defined Service. (The New button is grayed out on root device).
Select a name from the Name drop-down list.
6.
Select the Enable session cache for service check box.
7.
Click OK to create a predefined service session cache.
8.
Click OK to save your changes.
9.
Configuring SIP Settings on page 116
Configuring Timeouts for Predefined Services (NSM Procedure) on page 115
Configuring MGCP Settings on page 118
Use the SIP Settings option to configure Session Initiation Protocol (SIP) as a service on
the security device. SIP is an Internet Engineering Task Force (IETF)-standard protocol
for initiating, modifying, and terminating multimedia sessions (such as conferencing,
telephony, or multimedia) over the Internet. SIP is used to distribute the session
description, to negotiate and modify the parameters of an existing session, and to
terminate a multimedia session.
The device can then screen SIP traffic, permitting or denying it based on a security policy
that you configure. SIP is a predefined service in ScreenOS and uses port 5060 as the
destination port. Security devices currently do not support NAT (network address
translation) with SIP.
SIP is used to distribute the session description and, during the session, to negotiate and
modify the parameters of the session. SIP is also used to terminate the session.
SIP messages consist of requests from client to server and responses to requests from
servers to clients with the purpose of establishing a session (or a call). A UA (user agent)
is an application that runs at the endpoints of the call and consists of two parts: the UAC
(user agent client) that sends SIP requests on behalf of the user, and a UAS (user agent
server) who listens to the responses and notifies the user when they arrive. Examples of
user agents are SIP proxy servers and SIP phones.
A call can have one or more voice channels. Each voice channel has two sessions (or two
media streams), one for RTP and one for RTCP. When managing the sessions, the security
device considers the sessions in each voice channel as one group. Settings such as the
inactivity timeout apply to a group as opposed to each session.
Setting SIP Inactivity Timeouts
Copyright © 2010, Juniper Networks, Inc.
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