Polycom SIP 3.1 manual For Outgoing Calls Invite Fallback

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Administrator’s Guide SoundPoint IP / SoundStation IP

SRV records will be tried, before falling back on A records if NAPTR and SRV records return no results. If no port is given, and none is found through DNS, 5060 will be used.

Refer to http://www.ietf.org/rfc/rfc3263.txt for an example.

Note

Failure to resolve a DNS name is treated as signalling failure that will cause a

 

failover.

 

 

Behavior When the Primary Server Connection Fails

For Outgoing Calls (INVITE Fallback)

When the user initiates a call, the phone will go through the following steps to connect the call:

1.Try to make the call using the working server.

2.If the working server does not respond correctly to the INVITE, then try and make a call using the next server in the list (even if there is no current registration with these servers). This could be the case if the Internet connection has gone down, but the registration to the working server has not yet expired.

3.If the second server is also unavailable, the phone will try all possible servers (even those not currently registered) until it either succeeds in making a call or exhausts the list at which point the call will fail.

At the start of a call, server availability is determined by SIP signaling failure. SIP signaling failure depends on the SIP protocol being used as described below:

If TCP is used, then the signaling fails if the connection fails or the Send fails.

If UDP is used, then the signaling fails if ICMP is detected or if the signal times out. If the signaling has been attempted through all servers in the list and this is the last server, then the signaling fails after the complete UDP timeout defined in RFC 3261. If it is not the last server in the list, the maximum number of retries using the configurable retry timeout is used. For more information, refer to Server <server/> on page A-7and Registration <reg/> on page A-107.

Warning If DNS is used to resolve the address for Servers, the DNS server is unavailable, and the TTL for the DNS records has expired, the phone will attempt to contact the DNS server to resolve the address of all servers in its list before initiating a call. These attempts will timeout, but the timeout mechanism can cause long delays (for example, two minutes) before the phone call proceeds “using the working server”. To mitigate this issue, long TTLs should be used. It is strongly recommended that an on-site DNS server is deployed as part of the redundancy solution.

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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 HttpsCertificate Authority List on page C-1 Modifying the Network ConfigurationMain Menu Dhcp Menu Server Menu Ethernet Menu Syslog Menu EM Power Name Possible Values Description Dhcp ClientDhcp Menu Phone IP AddressPossible Name Values Description MenuSection, 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 unless you want to disable the PC port SettingEthernet 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 PhonesSIP/ on page A-10 Configuration on page A-4PhoneMACaddress.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 Configuring SoundPoint IP / SoundStation IP Phones Locally Setting Up Basic FeaturesThis chapter also provides instructions on Administrator’s Guide SoundPoint IP / SoundStation IP Call Log Call TimerCall Waiting Calling Party Identification Called 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 LocalHandset, Headset, and Speakerphone Do Not Disturb116 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 names7000, the maximum speed-dial index is Local Digit MapAuto-reject Soft Key Activated User Interface Microphone MuteSpeed Dial Boot server Ethernet Time and Date DisplayEthernet Switch Idle Display AnimationTheir phone Graphic Display BackgroundsYour 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 Corporate Directory Network Address TranslationNat/ on page A-120 Settings Basic Preferences Corporate Directory View This section contains the following information 670 have a functioning USB port Recording and Playback of Audio CallsDisplay 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 promptsSpecial Characters VersionVersion 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 stringContact Directory Changes Using Call Park KeyWell 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 ConferenceSoftkey.feature.newcall = Update the sip.cfg configuration as follows103 Update sip.cfg as follows Voice Mail Integration Multiple Registrations Server server/ on page A-7Server Redundancy Automatic Call DistributionServer/ on page A-7, and Registration reg/ on page A-107 DNS SIP Server Name Resolution For Outgoing Calls Invite Fallback Configured Phone ConfigurationPhone 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 the reg.x.server.y.address to the LCS server name Set reg.x.auth.password to the LCS passwordLocate 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-28Following table summarizes the phone’s audio codec support Audio CodecsEffective Comfort Noise Fill Background Noise SuppressionOn page A-38 and Codec Profiles audioProfile/ on page A-41 Automatic Gain Control IP Type-of-ServiceIeee 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 ProtocolConfiguration on page A-124 Configuring SoundPoint IP / SoundStation IP Phones LocallyDevice.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 fileReading an Application Log Testing Phone HardwareFollowing 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 Ept = 325,326,327,328,329,330 This attribute also includesLcl.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 Following table, x is the sampled audio file number Sound Effects seTo SoundPointIPWelcome.wav Patterns pat Ring type rt Instruction Meaning ExampleCall Progress Patterns Miscellaneous PatternsCall progress Use within phone Pattern number Ringer Patterns Ringer pattern number Default descriptionCall progress Pattern number Use within phone Miscellaneous Patterns Miscellaneous Pattern number Use within phonePatterns on page A-34 SequentialDefined 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 If voice.vadEnable is set to 0, add attribute line Voice.vadEnable parameterCentral Report Collector collector Alert Reports alert Central Report Collector collector Server server RTCP-XR rtcpxrNable.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.filterByPortWeb Server httpd Default value is usedValue 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 previewFollowing indicators are used by the phone Indicators indBitmaps bitmap IP500/, IP600/, IP4000/, Platform IP300/, IP 330/, IP400IP7000/ on page A-80 IP300/, IP330/, IP400/, IP500/, IP600 Attribute Permitted Interpretation ValuesIP4000/, and IP7000/ tag above LEDs led Following table, x is the LED numberEvent Logging log Level InterpretationTwo types of logging are supported Three formats are available for the event timestampType Example Syslog Menu on Log.render.level maps toYou 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 ramdiskRequest request Delay delayValue Feature feature Resource res Finder finder Quotas quotasValue. For the SoundStation IP 6000 Phones, this value is internally replaced by 2XPhones, 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 phonesFunction is selected If mb.main.idleTimeoutUsed. Refer to User Preferences up/ on Applications apps Detrimental effect on performance of the phoneNon-Null values Apps.push.password must be set to non-NullValues DNS Cache dns Peer Networking pnetNaptr 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-7Shared line counts as a call for every phone Refer to Call Handling Configuration callSharing that registration If reg.x.serverFeatureControl.cf is not Calls call VoIpProt.SIP.strictLineSeize isMore information, refer to SIP SIP/ on Set to 1 enabled, this parameter is ignored. ForSylantro call server only If call.missedCallTracking.x.enabled is Parameter is enabled Forwarding is enabled, thisDiversion 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 attendantValue 0 if the call server is Microsoft Live Roaming Buddies roamingbuddiesCommunications 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 Render/ on page A-86 Refer to Basic Logging level/change/Menu on Administrator’s Guide SoundPoint IP / SoundStation IP Session Initiation Protocol SIP RFC and Internet Draft Support Following SIP request messages are supported Request SupportMethod Supported Following SIP request headers are supported Header SupportHeader Supported Header Supported Following SIP responses are supported Response SupportResponse 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 DepthColor RGB Values Decimal Hexadecimal RGB ValuesModel 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 NumberDisabling PC Ethernet Port Select Save ConfigPress 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 Electronic Hookswitch Graphic Display Backgrounds New or Changed FeaturesDistribution Zip File Backlight Intensity Configuration File ChangesMetrics 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.