Network Address Translation nat | Polycom 1725-11530-200 Rev A1

Administrator’s Guide - SoundPoint IP / SoundStation® IP

Optimization

4.6.2.6 Network Address Translation <nat/>

These parameters define port and IP address changes used in NAT traversal. The port changes will change the port used by the phone, while the IP entry simply changes the IP advertised in the SIP signaling. This allows the use of simple NAT devices that can redirect traffic, but do not allow for port mapping. For example, port 5432 on the NAT device can be sent to port 5432 on an internal device, but not port 1234.

	Permitted
Attribute	Values	Default	Interpretation

nat.ip	dotted-deci-	Null	IP address to advertise within SIP signaling -
	mal IP		should match the external IP address used by the
	address		NAT device.

nat.signalPort	1024 to	Null	If non-Null, this port will be used by the phone for
	65535		SIP signaling, overriding the value set for voIp-
			Prot.local.signalPort in sip.cfg.

nat.mediaPortStart	1024 to	Null	If non-Null, this attribute will be used to set the
	65535		initially allocated RTP port, overriding the value
			set for tcpIpApp.port.rtp.mediaPortRangeStart in
			sip.cfg. Refer to 4.6.1.10.3.1 RTP <RTP/> on
			page 124.

nat.keepalive.inter-	0 to 3600	Null	If non-Null (or 0), the keepalive interval in sec-
val			onds. This parameter is used to set the interval at
			which phones will send a keep-alive packet to the
			gateway/NAT device to keep the communication
			port open so that NAT can continue to function as
			setup initially.
			The Microsoft® Office Live Communications
			Server 2005 keepalive feature will override this
			interval. If you want to deploy phones behind a
			NAT and connect them to Live Communications
			Server, the keepalive interval received from the
			Live Communications Server must be short
			enough to keep the NAT port open. Once the TCP
			connection is closed, the phones stop sending
			keep-alive packets.

160	Copyright © 2006 Polycom, Inc.

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Polycom 1725-11530-200 Rev A1 manual Network Address Translation nat

Contents

Administrator’s Guide SoundPoint/SoundStation IP SIP Polycom, Inc Table of Contents Call Management Features Optimization Configuration Files Session Initiation Protocol SIP 163 Viii Copyright 2006 Polycom, Inc Overview Copyright 2006 Polycom, Inc Installation and Operation Installation Models Installation Process Basic Network Setup Dhcp or Manual TCP/IP SetupPhones boot up in two phases Configuration File Local Parameter Provisioning File TransferSubnet mask IP gateway Sntp GMT offset Specified 300 301, 430 Protocol600 4000 Local User Interface Setup Menus Main MenuName Possible Values a Description Dhcp Menu Name Possible Description Values Possible Name Values Description Server Menu Name Possible Values Description Reset to Factory Defaults Ethernet Menu Application Configuration Centralized ConfigurationConfiguration Files Master configuration files contain six XML attributes Default Master Configuration File Application Configuration Files Category Description Example Setting Flash Parameters from Configuration Files Name Possible Values Description Name Possible Values Description Boot Server Deployment for the Phones Step Instructions Account-by-account basis Step Instructions Local Phone Configuration Passwords Management of File Encryption and Decryption Changing the Key on the Phone Copyright 2006 Polycom, Inc Page Basic Features Call TimerCall Log Phone maintains a call log. The log Call Waiting Called Party IdentificationCalling Party Identification Missed Call Notification Configurable Feature Keys SoundPoint IP 300 and 301 Key Layout SoundPoint IP 430 Key Layout SoundPoint IP 500 and 501 Key Layout SoundPoint IP 600 and 601 Key Layout SoundPoint IP 4000 Key Layout Key ID IP 300 IP 500 IP 600Function Connected Party Identification Key ID IP 300 Context Sensitive Volume Control Customizable Audio Sound Effects Administration Distinctive Incoming Call Treatment Message Waiting IndicationDistinctive Incoming Call Treatment Distinctive Ringing Distinctive Call Waiting Do-Not-Disturb Handset, Headset, and Speakerphone Local Contact Directory Local Contact Directory File Format Local Contact Directory File exampleElement Permitted Values Interpretation Local Digit Map Microphone Mute Multiple Line Keys per Registration Multiple Call Appearances Shared Call Appearances Line-seize subscription for mutual exclusion feature Bridged Line Appearances Figuration files or the local phone user interface should be Customizable Fonts and Indicators Busy Lamp Field Soft Key-Driven User Interface Speed Dial Time and Date Display Set the time and date display formats Call Management Features Automatic Off-hook Call PlacementIdle Display Animation Call Hold Call Transfer Three-Way Conference, Local or Centralized Call Diversion Call Forward Directed Call Pick-up Group Call Pick-up Call Park / Retrieve Audio Processing Features Low-Delay Audio Packet TransmissionJitter Buffer and Packet Error Concealment Last Call Return Voice Activity Detection Acoustic Echo Cancellation AEC Dtmf Tone GenerationDtmf Event RTP Payload Audio Codecs Following table summarizes the phone’s audio codec supportSample Effective Algorithm Presence and Instant Messaging Features Background Noise Suppression BNSComfort Noise Fill Automatic Gain Control AGC Localization Features Instant MessagingMultilingual User Interface Name Range +FF00 U+FFFF Downloadable Fonts Synthesized Call Progress Tones Advanced Server Features Voice Mail Integration SoundPoint IP 500, 501, 600 and 601 instant mes Multiple Registrations Local ACD login / logout ACD agent available / unavailableServer Redundancy DNS SIP Server Name Resolution Microsoft Office Live Communications Server 2005 Integration Central Configuration File Changes Modify the sip.cfg configuration file as followsModify the phone1.cfg configuration file as follows Optional Modify the sip.cfg configuration file as follows Accessory Internet Features Security FeaturesLocal User and Administrator Privilege Levels MicroBrowser Incoming Signaling Validation Custom Certificates Configuration File Encryption Administrator’s Guide SoundPoint IP / SoundStation IP Ethernet Switch Application Network SetupSoundPoint IP Switch Port Priorities Real-Time Transport Protocol Ports Working with Network Address Translation Updating and Rebooting Manual Reboot Menu Option or Key Presses Centralized RebootPeriodic Polling For Upgrades Event Logging Audio Quality Issues and VLANs IP TOSLog files uploaded in this manner are named Ieee 802.1p/Q Rtcp Support Configuration Files SIP Configuration sip.cfg Protocol volpProt Local localServer server Permitted Attribute Values Permitted Attribute Values TLS SDP SDP SIP SIP Attribute Values Attribute Values Permitted Attribute Values Default Interpretation Invite Default Interpretation Request Validation requestValidation Attribute Conference Setup conference Attribute Default Interpretation Ues Attribute Permitted Values Default Interpretation Dial Plan dialplanDigit Map digitmap Routing routing Localization localization Emergency emergency Multilingual multilingual Permitted Attribute Values Interpretation Date and Time datetime Adding New Languages User Preferences userpreferences Attribute Permitted Default Interpretation Values Tones tonesDual Tone Multi-Frequency Dtmf All three blocks use the same chord set specification format Chord Sets chordsets Following sampled audio Wave file .wav formats are supported Attribute Permitted Interpretation Values Following table, x is the sampled audio file number Attribute Permitted Values Interpretation Sound Effects soundeffects Patterns patternsInstruction Meaning Example 104 Copyright 2006 Polycom, Inc Ringer pattern Number Default description Call Progress PatternsCall progress Use within phone Pattern number Ringer Patterns Miscellaneous Patterns Miscellaneous Use within phone Pattern number Ring type ringType Miscellaneous Pattern number Use within phone Voice Settings voice Voice Coding Algorithms codecsFollowing voice codecs are supported Codec Profiles profiles Volume Persistence volume Default Interpretation Values Attribute Default Gains gains 112 Copyright 2006 Polycom, Inc Acoustic Echo Cancellation AEC Acoustic Echo Suppression AES Background Noise Suppression NS Automatic Gain Control AGC Receive Equalization Rxeq Transmit Equalization Txeq Quality of Service QOS Voice Activity Detection VADEthernet Ieee 802.1p/Q Ethernet IP TOS IP Call Control CallControlOther Other 120 Copyright 2006 Polycom, Inc Attribute Permitted Values Default Basic TCP/IP TcpipNetwork Monitoring netMon Time Synchronization Sntp 122 Copyright 2006 Polycom, Inc Copyright 2006 Polycom, Inc 123 1.10.3 Port port 1.10.3.1 Call Handling Configuration call Configuration cfgWeb Server Httpd Shared Calls shared Attribute Values Default Interpretation Hold, Local Reminder hold/localReminder Directory directory Presence presence Fonts font 501 Example of use 1.15.1 IP400 font IP400 1.15.2 IP500 font IP5001.15.3 IP600 font IP600 Keys keys Following table lists the functions that are available Bitmaps bitmaps Platform IP300/, IP400/, IP500/, IP600/ and IP4000Bitmaps used by the phone are defined in this section Indicators indicators Classes Classes Assignments Assignments Following table, x is the LED number LEDs ledGraphic Icons gi/ IP300/, IP400/, IP500/, IP600/ and IP4000 Event Logging logging Level Interpretation Basic Logging level/change/ and render Three formats are available for the event timestampType Example Copyright 2006 Polycom, Inc 139 These settings affect security aspects of the phone Security securityScheduled Logging Parameters scheduled Password Lengths pwd/length Encryption encryption Provisioning provisioning RAM Disk RAMdisk Delay delay Request request Feature feature Resource resource Finder finderQuotas quotas MicroBrowser microbrowser Idle Display idleDisplay Main Browser main Browser Limits limits 148 Copyright 2006 Polycom, Inc Per-phone Configuration phone1.cfg Registration reg 150 Copyright 2006 Polycom, Inc Copyright 2006 Polycom, Inc 151 152 Copyright 2006 Polycom, Inc Automatic Off-hook Call Placement autoOffHook Calls callDo Not Disturb donotdisturb Missed Call Configuration serverMissedCall Diversion divert Calls can be automatically diverted when the phone is busy Forward All fwdBusy busy No Answer noanswer Do Not Disturb dnd Copyright 2006 Polycom, Inc 157 158 Copyright 2006 Polycom, Inc Message Waiting Indicator mwi Messaging msg Network Address Translation nat Attendant attendant Roaming Buddies roamingbuddies Roaming Privacy roamingprivacy Basic Protocols RFC and Internet Draft SupportRequest Support Title Header Support Header Supported Copyright 2006 Polycom, Inc 165 Response Support 4.1 1xx Responses Provisional4.2 2xx Responses Success Response Supported 4.3 3xx Responses Redirection 4.4 4xx Responses Request Failure 4.5 5xx Responses Server Failure Hold Implementation Reliability of Provisional ResponsesTransfer Third Party Call Control Protocol Extensions Phone supports the following SIP protocol extensions Shared Call Appearance Signaling Or their successors Bridged Line Appearance Signaling Dialog for bridged line appearance subscribe and notify Appendix Trusted Certificate Authority List 174 Copyright 2006 Polycom, Inc Miscellaneous Administrative Tasks Adding a Background LogoModel Width Height Color Depth RGB Values Color Decimal Configuration File Changes Third Party Software Attribution 178 Copyright 2006 Polycom, Inc Copyright 2006 Polycom, Inc 179 Curl Copyright and Permission Notice