Polycom SIP 3.1 manual Following voice codecs are supported, These codecs include

Page 192

Administrator’s Guide SoundPoint IP / SoundStation IP

Voice Coding Algorithms <codecs/>

The following voice codecs are supported:

 

 

 

 

Sample

 

Effective Audio

Algorithm

MIME Type

Label

Bit Rate

Rate

Frame Size

Bandwidth

 

 

 

 

 

 

 

G.711μ-law

PMCU

G711mu

64 Kbps

8 Ksps

10ms - 80ms

3.5 KHz

 

 

 

 

 

 

 

G.711a-law

PCMA

G711A

64 Kbps

8 Ksps

10ms - 80ms

3.5 KHz

 

 

 

 

 

 

 

G.722

G722/8000

G722

64 Kbps

16 Ksps

10ms - 80ms

7 KHz

 

 

 

 

 

 

 

G.722.1

G7221/16000

G7221

16 Kbps,

16 Ksps

20ms - 80ms

7 KHz

 

 

 

24 Kbps,

 

 

 

 

 

 

32 Kbps

 

 

 

 

 

 

 

 

 

 

G.722.1C

G7221/

G7221C

24 Kbps

32 Ksps

20ms - 80ms

14 KHz

 

32000

 

32 Kbps

 

 

 

 

 

 

48 Kbps

 

 

 

 

 

 

 

 

 

 

G.729AB

G729

G729AB

8 Kbps

8 Ksps

10ms - 80ms

3.5 KHz

 

 

 

 

 

 

 

Lin16

L16/16000

L16

25.6 Kbps

16 Ksps

10ms

7 KHz

 

L16/32000

 

51.2 Kbps

32 Ksps

 

14 KHz

 

L16/48000

 

76.8 Kbps

48 Ksps

 

22 KHz

 

 

 

 

 

 

 

Siren14

SIREN14/

SIREN14

24 Kbps

32 Ksps

20ms - 80ms

14 KHz

 

16000

 

32 Kbps

 

 

 

 

 

 

48 Kbps

 

 

 

 

 

 

 

 

 

 

Siren22

SIREN22/

SIREN22

32 Kbps

48 Ksps

20ms - 80ms

22 KHz

 

48000

 

48 Kbps

 

 

 

 

 

 

64 Kbps

 

 

 

 

 

 

 

 

 

 

These codecs include:

Codec Preferences <codecPref/>

Codec Profiles <audioProfile/>

Codec Preferences <codecPref/>

Attribute

Permitted

Default

Interpretation

Values

 

 

 

 

voice.codecPref.G711Mu

Null, 1-3

1

Specifies the codec preferences for

 

 

 

SoundPoint IP 320, 330, 430, 500, 501,

voice.codecPref.G711A

 

2

 

600 and 601 platforms.

 

 

 

voice.codecPref.G729AB

 

3

1 = highest

 

 

 

3 = lowest

 

 

 

Null = do not use

 

 

 

Give each codec a unique priority, this will

 

 

 

dictate the order used in SDP negotiations.

 

 

 

 

A - 38

Image 192
Contents SIP Disclaimer Copyright NoticeAbout This Guide Administrator’s Guide SoundPoint IP / SoundStation IP Contents Administrator’s Guide SoundPoint IP / SoundStation IP Contents Troubleshooting Your SoundPoint IP / SoundStation IP Phones Contents Administrator’s Guide SoundPoint IP / SoundStation IP Introducing the SoundPoint IP / SoundStation IP Family SoundPoint IP Desktop PhonesAdministrator’s Guide SoundPoint IP / SoundStation IP SoundPoint IP SoundPoint IP 550/560 SoundPoint IP 600/601 SoundStation IP Conference Phones Currently supported conference phones are SoundStation IP Key Features of Your SoundPoint IP / SoundStation IP Phones SoundPoint IP 600 Administrator’s Guide SoundPoint IP / SoundStation IP Overview Where SoundPoint IP / SoundStation IP Phones Fit SoundPoint IP / SoundStation IP Phones onBootROM Session Initiation Protocol Application ArchitectureApplication Configuration Master Configuration Files Application Configuration FilesApplication Configuration Files Resource Files Available Features Ring tones Synthesized tones Contact directoriesOverview Microsoft Live Communications Server Overview Administrator’s Guide SoundPoint IP / SoundStation IP New Features in SIP Administrator’s Guide SoundPoint IP / SoundStation IP Setting up Your System Setting Up the Network Dhcp or Manual TCP/IP SetupFor more information on Dhcp options, go to Supported Provisioning Protocols FTP Tftp Http HttpsModifying the Network Configuration Certificate Authority List on page C-1Main Menu Dhcp Menu Server Menu Ethernet Menu Syslog Menu EM Power Name Possible Values Description Dhcp ClientDhcp Menu Phone IP AddressMenu Possible Name Values DescriptionSection, Server Menu Server Menu MenuDhcp on page C-23 Name Possible Values DescriptionServer. Refer to Supported Provisioning Protocols on Or later. Passive FTP is still supportedPassword these characters if they are correctly escaped Using the method specified in RFCPassword, this will be ignored This will be ignoredSetting Setting unless you want to disable the PC portEthernet menu Setting Up the Boot Server Refer to Basic Logging level/change/ and renderInformation, contact your Certified Polycom Reseller Create account and home directoryEach phone may open multiple connections to the server These permissions, but will not be able to upload filesDeploying Phones From the Boot Server You must decide on a boot server security policySip.cfg Phone1.cfg 000000000000.cfg Directory~.xml SoundPointIP-dictionary.xmlone of each supported languageConfiguration Files on page C-4 Provisioning PhonesConfiguration on page A-4 SIP/ on page A-10PhoneMACaddress.cfg 5EL@ Provisioning SoundStation IP 7000 Phones Using CLink Upgrading SIP Application Supporting SoundPoint IP and SoundStation IP PhonesSupporting SoundPoint IP 300 and 500 Phones To upgrade your SIP application Administrator’s Guide SoundPoint IP / SoundStation IP Setting Up Basic Features Configuring SoundPoint IP / SoundStation IP Phones LocallyThis chapter also provides instructions on Administrator’s Guide SoundPoint IP / SoundStation IP Call Timer Call LogCall Waiting Called Party Identification Calling Party IdentificationMissed Call Notification Context Sensitive Volume Control Central boot serverConnected Party Identification Customizable Audio Sound EffectsMessage Waiting Indication Distinctive Incoming Call TreatmentMessages and voice messages are waiting Saf/ on page A-30 or Sound Effects se/ on page A-31Distinctive Ringing Distinctive Call WaitingAddress-directory Xml LocalDo Not Disturb Handset, Headset, and Speakerphone116 Local Contact Directory Userpreferences/on page A-107Direct Ory.xmlXml ?xml version=1.0 encoding=UTF-8 standalone=yes ? directoryDirectory Element Permitted Values InterpretationUTF-8’s variable length encoding Space is added between first and last namesLocal Digit Map 7000, the maximum speed-dial index isAuto-reject Microphone Mute Soft Key Activated User InterfaceSpeed Dial Boot server Ethernet Time and Date DisplayEthernet Switch Idle Display AnimationGraphic Display Backgrounds Their phoneYour choice Automatic Off-Hook Call Placement Call HoldFor images, select a filename. For example AutoOffHook/ on page A-112Call Transfer Hold/localReminder/ on page A-67Local / Centralized Conferencing Manage ConferencesCall Forward Directed Call Pick-Up Setting Up Advanced Features Group Call Pick-UpCall Park/Retrieve Last Call ReturnConfiguring Your System Configurable Feature Keys Feature Key Layouts on page C-12Multiple Line Keys per Registration Multiple Call AppearancesShared Call Appearances Bridged Line Appearance Busy Lamp Field Customizable Fonts and Indicators EchnicalBulletinspub.htmlLive Communications Server 2005 Integration on Attendant.uriCentral boot Instant MessagingMultilingual User Interface Server Sip.cfgSwedish Fonts, refer to Fonts font/ on page A-72Synthesized Call Progress Tones MicrobrowserSaf/ on page A-30 Call Progress Patterns on page A-33Real-Time Transport Protocol Ports Network Address Translation Corporate DirectoryNat/ on page A-120 Settings Basic Preferences Corporate Directory View This section contains the following information Recording and Playback of Audio Calls 670 have a functioning USB portDisplay Daisy-Chaining Phones Provisioning Phones Over CLink Enhanced Feature Keys Efk Efklist Efkprompt Version Special CharactersThis element describes behavior of enhanced feature key EfkEfklist This element contains the following parametersEfkprompt This element describes the behavior of the user promptsVersion Special CharactersVersion efk.version=2 Macro Action Prompt Macro Substitution Expanded Macros Macro ActionUsing Invite if no active call or Dtmf if an active Prompt Macro SubstitutionCall. The use of refer method is call server Dependentand may require the addition of star codesCollected. The macros are case sensitive Prompt is not required for every macroExpanded Macros Contact Directory File Format onExamples Configuration File Changes Enhanced Feature Key XML FilesAction String Example Action stringUsing Call Park Key Contact Directory ChangesWell as others mapped to Park Return and Call Pickup Configurable Soft Keys New Call End Call Split Join Forward MyStatus and Buddies Hold, Transfer, and ConferenceUpdate the sip.cfg configuration as follows Softkey.feature.newcall =103 Update sip.cfg as follows Voice Mail Integration Multiple Registrations Server server/ on page A-7Automatic Call Distribution Server RedundancyServer/ on page A-7, and Registration reg/ on page A-107 DNS SIP Server Name Resolution For Outgoing Calls Invite Fallback Phone Configuration ConfiguredPhone Operation for Registration Presence Boot server Address-directory Microsoft Live Communications Server 2005 IntegrationImmediately with business contacts Examples onRoamingbuddies/ on page A-122 Roamingprivacy/ on page A-123Refer to Roaming Buddies roamingbuddies/ on page A-122 Refer to Roaming Privacy roamingprivacy/ on page A-123Set reg.x.auth.password to the LCS password Set the reg.x.server.y.address to the LCS server nameLocate the roamingprivacy attribute Access URL in SIP Message Web Content Examples User Interface Signaling ChangesWeb Content Status Indication Settings Menu Static DNS Cache Example Dns.cache.A.1 , dns.cache.A.2 , and so on Set to null to force SRV lookups Display of Warnings from SIP Headers Setting Up Audio Features Low-Delay Audio Packet Transmission Jitter Buffer and Packet Error ConcealmentVoice Activity Detection Dynamic Noise Reduction Treble/Bass ControlsAcoustic Echo Cancellation Dtmf Tone GenerationDtmf Event RTP Payload DTMF/ on page A-28Audio Codecs Following table summarizes the phone’s audio codec supportEffective Background Noise Suppression Comfort Noise FillOn page A-38 and Codec Profiles audioProfile/ on page A-41 IP Type-of-Service Automatic Gain ControlIeee 802.1p/Q Three types of quality reports can be enabled Periodic-Generated during a call at a configurable periodVoice Quality Monitoring ThresholdSetting Up Security Features Dynamic Noise ReductionTreble/Bass Controls Monitoring/ on page A-52Local User and Administrator Privilege Levels Custom CertificatesPwd/length/ on page A-89 Tinspub.htmlIncoming Signaling Validation Configuration File EncryptionConfiguration changes can performed locally Secure Real-Time Transport ProtocolConfiguring SoundPoint IP / SoundStation IP Phones Locally Configuration on page A-124Device.cfg Passwords Troubleshooting Your SoundPoint IP / SoundStation IP Phones Error Messages BootROM Error MessagesApplication Error Messages Status Menu Log Files Scheduled Logging Manual Log Upload Reading a Boot Log Following figure shows a portion of a boot log fileTesting Phone Hardware Reading an Application LogFollowing figure shows a portion of an application log file Power and Startup Symptom Problem Corrective ActionControls Access to Screens and Systems To Rebooting the Phone onCalling Displays Phone on page C-10Audio UpgradingOice/soundpointip/VoIPTechnicalBullet Inspub.htmlAdministrator’s Guide SoundPoint IP / SoundStation IP Configuration Files Master Configuration Files One will cause a reboot loop Application Configuration CONFIGFILES=phone1MACADDRESS.cfg, sip.cfg MISCFILES=Configuration Files This configuration attribute is defined as follows Attribute Permitted Default Interpretation ValuesProtocol voIpProt This attribute includesPermitted Attribute Values Default Interpretation If voIpProt.server.x.address is aVoIpProt.server.x.transport is set to If voIpProt.server.x.transport is set toVoIpProt.server.x.address is an IP VoIpProt.SIP.lcs Parameter if set to 1 when the parameter To 1 default isPermitted Attribute Values Default Interpretation Reg.x.auth.optimizedInFailover takes This attribute also includes Ept = 325,326,327,328,329,330Lcl.ml.lang.tags.x in Multilingual ml Outbound Proxy outboundProxy Due to the additional signaling required Alert Information alertInfoRequest Validation requestValidation May have a negative performance impactConference Setup conference Special Events specialEventSupported when configured with the values Dial Plan dialplanUDP, TCP, or TLS Dialplan.applyToCallListDialThis attributes also includes Digit Map digitmap Routing routingConfiguration Files Attribute Permitted Values Default Interpretation Server server Emergency emergencyLocalization lcl Server serverEmergency emergency Multilingual ml Date and Time datetimeAttribute Permitted Values Interpretation Lcl.ml.lang.menu.1Lcl.ml.lang.menu.2 Lcl.ml.lang.menu.3Lcl.datetime.date.longFormat Lcl.datetime.date.dateTopLcl.ml.lang.tags.1 = Zh-cn,zhq=0.9,enq=0.8Optional Set lcl.ml.lang to be the new languageregion string User Preferences up Permitted Attribute Values InterpretationOnIntensity, it will be replaced with OnIntensity valueTones tones Dual Tone Multi-Frequency Dtmf Chord-Sets chordOnly be enabled when tone.dtmf.viaRtp is DisabledBe enabled when tone.dtmf.viaRtp is Sampled Audio for Sound Effects saf Sound Effects se Following table, x is the sampled audio file numberTo SoundPointIPWelcome.wav Patterns pat Ring type rt Instruction Meaning ExampleMiscellaneous Patterns Call Progress PatternsCall progress Use within phone Pattern number Ringer pattern number Default description Ringer PatternsCall progress Pattern number Use within phone Miscellaneous Patterns Miscellaneous Pattern number Use within phoneSequential Patterns on page A-34Defined in Call Progress Patterns on page A-33 Voice Settings voice Following voice codecs are supported These codecs includeCodec Preferences codecPref Codec Profiles audioProfile Codec Preferences codecPrefPermitted Attribute Values Default Interpretation Voice.codecPref.IP7000.G722 Codec Profiles audioProfile Attribute Default Attribute Default Attribute Default Acoustic Echo Cancellation aec Acoustic Echo Suppression aes Background Noise Suppression ns Feature Receive Equalization rxEq Transmit Equalization txEq Attribute Default Voice.vadEnable parameter If voice.vadEnable is set to 0, add attribute lineCentral Report Collector collector Alert Reports alert Server server RTCP-XR rtcpxr Central Report Collector collectorNable.periodic is set 1, since Alert Reports alert Quality of Service QOS Following settings control the 802.1p/Q userpriority fieldRTCP-XR rtcpxr Ethernet Ieee 802.1p/Q ethernet IP TOS IPThese parameters apply to RTP packets Call Control callControlOther other RTP rtp Call Control callControlRTP rtp Basic TCP/IP Tcpip Qos.ip.callControl…Attribute Permitted Default Values Permitted Attribute Values Default Interpretation If fixedDayEnable is set to Start.dayOfWeekStart.date is ignored Stop.dayOfWeekMust be enabled for this to work RTP rtpTcpIpApp.port.rtp.filterByIp TcpIpApp.port.rtp.filterByPortDefault value is used Web Server httpdValue that is out of range, Call Handling Configuration call Configuration cfgReg.x.callsPerLineKey. Refer to Registration If call.stickyAutoLineSeize is set to 1, thisShared Calls shared Hold, Local Reminder hold/localReminder SoundPoint IP 330/320 onlyBroadWorks calls server only. You must change Value if your organization uses a different callServers IP 4000, 6000, and 7000 phones. For otherPhones a quick press and release of the line Key will resume a call whereas pressingDirectory dir Local Directory local Corporate Directory corpDir.local.volatile.4meg Dir.local.volatile.8meg, thisSoundPoint IP 320/330 is disabled Read only, speed dial entry onEnter the speed dial index followed By #Used for display purposes only Dir.corp.viewPersistence 600, and 601 legacy phones, useLeg tagged parameter. This Prevents slow behavior after exitingPresence pres Fonts fontSoundPoint IP 320, 330, 430, 500 SoundPoint IP 550, 560, 600, 601, 650,IP330 font IP330 This configuration attribute is defined as follows Keys keyFollowing table lists the functions that are available FunctionsBuilt-in default solid pattern is displayed Backgrounds bgSame to display correctly on grayscale Individual phone when the user lightens or Darkens the graphic during previewIndicators ind Following indicators are used by the phoneBitmaps bitmap Platform IP300/, IP 330/, IP400 IP500/, IP600/, IP4000/,IP7000/ on page A-80 Attribute Permitted Interpretation Values IP300/, IP330/, IP400/, IP500/, IP600IP4000/, and IP7000/ tag above LEDs led Following table, x is the LED numberEvent Logging log Level InterpretationThree formats are available for the event timestamp Two types of logging are supportedType Example Log.render.level maps to Syslog Menu onYou do not change this value Set starting with log.sched.x where x identifies the task Support append mode unlessServer is set up for this Uploaded if no new events haveEncryption encryption Password Lengths pwd/length Security secLicense license Provisioning prov RAM Disk ramdiskDelay delay Request requestValue Feature feature Resource res Finder finder Quotas quotasPhones, this value is internally replaced by 2X Value. For the SoundStation IP 6000Phones, this value is internally replaced by 4X Microbrowser mb SoundStation IP 4000, 6000, and 7000 phonesThis value is internally replaced by 2X the value. For Replaced by 4X the valueMiscellaneous XML errors can occur on SoundPoint IP 430, 501, 550, 560, 600650, and 670 and SoundStation IP 4000 7000 phonesIf mb.main.idleTimeout Function is selectedUsed. Refer to User Preferences up/ on Applications apps Detrimental effect on performance of the phoneApps.push.password must be set to non-Null Non-Null valuesValues Peer Networking pnet DNS Cache dnsNaptr NAPTR/ attribute SRV SRV Http//tools.ietf.org/html/rfc2915 Http//tools.ietf.org/html/rfc2782 Soft Keys softkey Macro Definition onPermitted Attribute Values Default Interpretation New Call and Callers soft keys For this soft key to be displayedPer-Phone Configuration Parameters includeRegistration reg User Preferences userpreferencesIs non-Null, all of the reg.x.server.y.xxx Parameters will override the parametersSpecified in sip.cfg in Server server/ on A-7Refer to Call Handling Configuration call Shared line counts as a call for every phoneSharing that registration If reg.x.serverFeatureControl.cf is not Calls call VoIpProt.SIP.strictLineSeize isSet to 1 enabled, this parameter is ignored. For More information, refer to SIP SIP/ onSylantro call server only If call.missedCallTracking.x.enabled is Forwarding is enabled, this Parameter is enabledDiversion divert Calls can be automatically diverted when the phone is busy Divert.x.contact will beEnabled, this parameter is Server-base call forwarding isDialplan.x.digitmap is not Plan dialplan/ on page A-17Dialplan.x.applyToUserDial When present, and if Digit Map digitmap/ on Message Waiting Indicator mwi Messaging msgServer/ on page A-118 Chosen. Refer to Voice Mail Integration onNetwork Address Translation nat VoIpProt.local.signalPort in sip.cfg Attendant attendantRoaming Buddies roamingbuddies Value 0 if the call server is Microsoft LiveCommunications Server Roaming Privacy roamingprivacy User Preferences userpreferencesVoIpProt.SIP.strictLineSeize VoIPProt.SIP.lineSeize.retries,Flash Parameter Configuration Enabled Setup onThis flash attributes are defined as follows For example, if device.net.ipAddress.set =Server address is preserved Refer to Basic Logging level/change/ Render/ on page A-86Menu on Administrator’s Guide SoundPoint IP / SoundStation IP Session Initiation Protocol SIP RFC and Internet Draft Support Request Support Following SIP request messages are supportedMethod Supported Header Support Following SIP request headers are supportedHeader Supported Header Supported Response Support Following SIP responses are supportedResponse Supported 3xx Responses Redirection 5xx Responses Server Failure Hold Implementation Reliability of Provisional ResponsesTransfer Third Party Call ControlShared Call Appearance Signaling Bridged Line Appearance SignalingMiscellaneous Administrative Tasks Trusted Certificate Authority ListAdministrator’s Guide SoundPoint IP / SoundStation IP Miscellaneous Administrative Tasks Encrypting Configuration Files Changing the Key on the Phone Option. This shows the digest fieldEncrypted and unencrypted file are the same Encryption/ on page A-89Adding a Background Logo Model Width Height Color DepthRGB Values Color RGB Values Decimal HexadecimalModel Associate Parameter Bitmaps IP300 IP300 IP330IP330 IP400 IP500 IP500BootROM/SIP Application Dependencies Animations IndicatorsModel BootROM SIP Application Migration DependenciesMultiple Key Combinations BootROM until the password prompt appears IP 4000 and 6000 6, 8 and * dial pad keysAbout three seconds IP 301 The two Line keys and the Up and Down arrow keysDefault Feature Key Layouts SoundStation IP 4000, 6000, and 7000 modelsSoundPoint IP Key ID FunctionSoundPoint IP 320/330 SoundPoint IP OPER0 14 # 12 11OPER Key ID SoundPoint IP 550/560/600/601/650/670 SoundStation IP Key ID Internal Key Functions Label Function LCR Label Function Assigning a Vlan ID Using Dhcp VLAN-A=10 VLAN-A=0x0a VLAN-A=012Parsing Vendor ID Information Miscellaneous Administrative Tasks Product, Model, and Part Number Mapping Product Name Model Name Product Part NumberSelect Save Config Disabling PC Ethernet PortPress Administrator’s Guide SoundPoint IP / SoundStation IP Third Party Software OpenSSL Third Party Software Zlib Copyright and Permission Notice Administrator’s Guide SoundPoint IP / SoundStation IP Index NumericsAdministrator’s Guide SoundPoint IP / SoundStation IP Dhcp IP TOS call control callControl A-58 IP400 font A-74Administrator’s Guide SoundPoint IP / SoundStation IP SDP SDP A-9 Sipsip A-10 POLYCOM, INC Application Programming Interface License API License Agreement for Development Purposes Support Services Export Controls Page Addendum to SIP 3.1 Administrator’s Guide New or Changed Features Electronic Hookswitch Graphic Display BackgroundsDistribution Zip File Configuration File Changes Backlight IntensityMetrics for listening and conversational quality Gains gain Receive Equalization rxEq Transmit Equalization txEqBackground bg Administrator’s Guide Addendum for the SoundPoint IP Multiple Key Combinations and Default Key Layout Key ID Administrator’s Guide Addendum for the SoundPoint IP
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SIP 3.1 specifications

Polycom SIP 3.1 is an advanced session initiation protocol designed to enhance voice and video communication in various business environments. As a pivotal component of Polycom’s telecommunication solutions, SIP 3.1 offers several features and characteristics that cater to the evolving needs of modern enterprises, particularly those that rely on seamless and efficient communication.

One of the standout features of Polycom SIP 3.1 is its robust interoperability. This protocol supports a wide range of endpoints and platforms, allowing organizations to integrate their existing systems with new technologies effortlessly. This flexibility ensures that businesses can leverage their previous investments while upgrading to the latest communication tools.

Another key aspect of Polycom SIP 3.1 is its enhanced call management capabilities. The protocol facilitates efficient call handling, enabling users to manage multiple calls seamlessly. Features like call hold, transfer, and conferencing are streamlined, which enhances productivity by allowing for efficient collaboration among team members. Additionally, it is optimized for low latency and high-quality audio, ensuring that conversations are clear and free from disruptions.

Security is paramount in today’s digital landscape, and Polycom SIP 3.1 addresses this concern with advanced encryption standards. By integrating security features such as Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP), it protects sensitive communication from unauthorized access and ensures that data remains confidential throughout the call.

Polycom SIP 3.1 also boasts compatibility with various video codecs, making it a versatile choice for video conferencing. This compatibility ensures high-quality video streams, which is essential for effective visual communication in remote meetings. Furthermore, the support for the H.264 codec provides efficient bandwidth usage, making high-definition video conferencing accessible, even in varying network conditions.

Moreover, the protocol provides strong support for presence and instant messaging, which enhances real-time communication among users. This integration of voice, video, and messaging capabilities fosters a more connected and collaborative work environment, allowing teams to engage effectively regardless of their geographical locations.

In summary, Polycom SIP 3.1 stands out as a sophisticated solution tailored to meet the demands of modern business communication. With its emphasis on interoperability, call management, security, video quality, and real-time collaboration, it caters to companies of all sizes seeking to optimize their communication infrastructure and enhance productivity in the workplace. As businesses continue to navigate the complexities of digital communication, Polycom SIP 3.1 remains a compelling choice in the market.