Sip headers – Telecom R & D

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SIP is a widely adopted application layer protocol used in VoIP calls and confernecing applciations and in IMS architeture or pure packet switched networks ... Skiptocontent SIPisawidelyadoptedapplicationlayerprotocolusedinVoIPcallsandconfernecingapplciationsandinIMSarchitetureorpurepacketswitchednetworks. MoreonSIP,itspacketstructure,transactionanddialogs,looseandstrictrecordrouting,locationservice,nearandfarendnating,andcommonlyusedSIPCallflowslikeRedirection,forking,clicktoDial–https://telecom.altanai.com/2013/07/13/sip-session-initiaion-protocol/(opensinanewtab) SIPRequestandRepsosnes TraditionalSIPheadersforCallsetupareINVITE,ACKandteardownareCANCELorBYE,howeverwithmoreadoptionnewermethodsspecifictoserviceswereaddedsuchas: MESSAGEMethodsforInstantMessagebasedservicesSUBSCRIBE,NOTIFYstandardisedbyEventnotificationextensionRFC3856PUBLISHtopushpresenceinformationtothenetwork OutliningtheSIPRequestsandResponsesintablesbelow, RequestMessage RequestMessageDescriptionREGISTERAClientusethismessagetoregisteranaddresswithaSIPserverINVITEAUserorServiceusethismessagetoletanotheruser/serviceparticipateinasession.Thebodyofthismessagewouldincludeadescriptionofthesessiontowhichthecalleeisbeinginvited.ACKThisisusedonlyforINVITEindicatingthattheclienthasreceivedafinalresponsetoanINVITErequestCANCELThisisusedtocancela pendingrequestBYEAUserAgentClientusethismessagetoterminatethecallOPTIONSThisisusedtoqueryaserveraboutitscapabilities ResponseMessage CodeCategoryDescription1xxProvisionalTherequesthasbeenreceivedandprocessingiscontinuing2xxSuccessAnACK,toindicatethattheactionwassuccessfullyreceived,understood,andaccepted.3xxRedirectionFurtheractionisrequiredtoprocessthisrequest4xxClientErrorTherequestcontainsbadsyntaxandcannotbefulfilledatthisserver5xxServerErrorTheserverfailedtofulfillanapparentlyvalidrequest6xxGlobalFailureTherequestcannotbefulfilledatanyserver SIPheaders Displaynames Fromoriginatorssipuri CSeqorCommandSequencecontainsanintegerandamethodname.TheCSeqnumberisincrementedforeachnewrequestwithinadialogandisatraditionalsequencenumber. Contact–SIPURIthatrepresentsadirectroutetotheoriginatorusuallycomposedofausernameatafullyqualifieddomainname(FQDN),alsoIPaddressesarepermitted.TheContactheaderfieldtellsotherelementswheretosendfuturerequests. Max-Forwards-tolimitthenumberofhopsarequestcanmakeonthewaytoitsdestination.Itconsistsofanintegerthatisdecrementedbyoneateachhop. Content Content-Type–descriptionofthemessagebody. Content-Type:application/h.323 Content-Type:message/sip Content-Type:application/sdp Content-Type:multipart/signed; protocol="application/pkcs7-signature"; micalg=sha1;boundary=boundary42 Content-Type:application/pkcs7-signature;name=smime.p7s ContentEncoding Content-Encoding:text/plain ContentLanguage Content-Language:en Content-Length–anoctet(byte)countofthemessagebody. Content-Disposition describeshowthemessagebodyor,formultipartmessages,amessagebodypartistobeinterpretedbytheUACorUAS.ItextendstheMIMEContent-Type DispositionTypes: “session”–bodypartdescribesasession,foreithercallsorearly(pre-call)media“render”–bodypartshouldbedisplayedorotherwiserenderedtotheuser.“icon”–bodypartcontainsanimagesuitableasaniconicrepresentationofthecallerorcallee“alert”–bodypartcontainsinformation,suchasanaudioclip Accept Accept–acceptableformatslikeapplication/sdporcurrency/dollars HeaderfieldwhereproxyACKBYECANINVOPTREG AcceptR-o-om*oAccept2xx---om*oAccept415-c-ccc AnemptyAcceptheaderfieldmeansthatnoformatsareacceptable. Accept-Encoding Accept-EncodingR-o-oooAccept-Encoding2xx---om*oAccept-Encoding415-c-ccc Accept-Language:languagesforreasonphrases,sessiondescriptions,orstatusresponsescarriedasmessagebodiesintheresponse. Accept-Language:da,en-gb;q=0.8,en;q=0.7 Accept-LanguageR-o-ooo Accept-Language2xx---om*o Accept-Language415-c-ccc Taggloballyuniqueandcryptographicallyrandomwithatleast32bitsofrandomness.identifyadialog,whichisthecombinationoftheCall-IDalongwithtwotags(fromToandFROMheaders) Call-Iduniquelyidentifyasession contact–sipurlalternativefordirectrouting Encryption Expires–whenmsgcontentisnolongervalid MandatorySIPheaders INVITEsip:[email protected]/2.0 Via:SIP/2.0/UDPhost.domain.com:5060 From:Bob To:Altanai Call-ID:[email protected] CSeq:1INVITE Informationalheaders Call-Infoadditionalinformationforexample,throughawebpage.The“card”parameterprovidesabusinesscard,forexample,invCard[36]orLDIF[37]formats.AdditionaltokenscanberegisteredusingIANA Call-Info:http://wwww.example.com/alice/photo.jpg;purpose=icon,http://www.example.com/alice/;purpose=info ContactContact:“Mr.Watson”;q=0.7;expires=3600,“Mr.Watson”[email protected];q=0.1m:;expires=60 Priorityindicatestheurgencyoftherequestasperceivedbytheclient.canhavethevalues“non-urgent”,“normal”,“urgent”,and“emergency”,butadditionalvaluescanbedefinedelsewhere Subject:Atornadoisheadingourway!Priority:emergency or Subject:WeekendplansPriority:non-urgent Subjectsummaryorindicatesthenatureofcall Subject:Needmoreboxess:TechSupport Supportedenumeratesalltheextensionssupported.cancontainlistofoptiontags,described Supported:100relk:100rel Unsupportedfeaturesnotsupported Unsupported:foo User-AgentinformationabouttheUACoriginatingtherequest. User-Agent:SoftphoneBeta1.5 OrganizationconveysthenameoftheorganizationtowhichtheSIPelementissuingtherequestorresponsebelongs. Organization:AltanaiTelecomCo. Warningadditionalinformationaboutthestatusofaresponse.Listofwarn-code 300Incompatiblenetworkprotocol:301Incompatiblenetworkaddressformats:302Incompatibletransportprotocol:303Incompatiblebandwidthunits:304Mediatypenotavailable:305Incompatiblemediaformat:306Attributenotunderstood:307Sessiondescriptionparameternotunderstood:330Multicastnotavailable:331Unicastnotavailable:370Insufficientbandwidth:399Miscellaneouswarning:1xxand2xxhavebeentakenbyHTTP/1.1. Warning:307isi.edu“Sessionparameter‘foo’notunderstood”Warning:301isi.edu“Incompatiblenetworkaddresstype‘E.164′” AutheticationandAuthorizationrelatedheaders Authentication-InfomutualauthenticationwithHTTPDigest.AUASMAYincludethisheaderfieldina2xxresponsetoarequestthatwassuccessfullyauthenticatedusingdigestbasedontheAuthorizationheaderfield. Authentication-Info:nextnonce=”47364c23432d2e131a5fb210812c” AuthorizationauthenticationcredentialsofaUA Authorization:Digestusername=”Alice”,realm=”atlanta.com”,nonce=”84a4cc6f3082121f32b42a2187831a9e”,response=”7587245234b3434cc3412213e5f113a5432″ Proxy-Authenticatecontainsanauthenticationchallenge. Proxy-Authenticate:Digestrealm=”atlanta.com”,domain=”sip:ss1.carrier.com”,qop=”auth”,nonce=”f84f1cec41e6cbe5aea9c8e88d359″,opaque=””,stale=FALSE,algorithm=MD5 Timers exponentialback-offonre-transmissions  SessionExpireHeaderFeild limitthetimeperiodoverwhichastatefulproxymustmaintainstateinformation.options Useragentsmustteardownthecallaftertheexpirationofthetimer,orallercansendre-INVITEstorefreshthetimer,enablinga“keepalive”mechanismforSIP. SDP(SessionDescriptionProtocol) SIPcanbearmanykindsofMIMEattachments,onesuchisSDP.Itisastandardforprotocoldefinitionforexchangeofmedia,metadataandothertransportrealtedattributesbetweentheparticpantsbeforeestablishingaVoIPcall. SDPsessiondescriptionisentirelytextualusingtheISO10646charactersetinUTF-8encodinganddescribedbyapplication/SDPmediatype. ItshouldbenotedthatSDPitselfdoesnotincorporateatransportprotocolandcanbeusedwithdifferenceprotoclslikeSessionannouncementproctols(SAP),SIP,HTTP,ElectronicMAIlMIMEextension,RTSPetc. IncaseofSIPSDPisencapsulatedinsideofSIPpacketanduseoffer/answermodeltoconveyinformationaboutmediastreaminmultimediasession. SDPbodycontains2parts:sessionbasedsectionstartingwithv=lineandmediabsesctionstartingwithm=lineMediaandTransportInformationcancontaintypeofmedialikevideo,audio,transportprotocollikeRTP/UDP/IP,H.320andformatofthemediasuchasH.261video,MPEGvideo,etc. SessionDescriptioninSDP protocolversion(v=)protocolversionmostlyversion0 originatorandsessionidentifier(o=) o= o=-64768885762848743442INIP4127.0.0.1 sessionname(s=)andsessioninformation(i=)sessionnameistextualandcancontainemptyspaceorevens=-butmustnotbeempty.Sessioninfomrationisoptionaltextualinformationaboutthesession URIofdescription(u=) EmailAddressandPhoneNumber(“e=”and“p=”) BothareoptionalfreetextstringSHOULDbeintheISO-10646charactersetwithUTF-8encoding NothethatifgiventhePhonenumbersSHOULDfollowinternationalpublictelecommunicationnumberspecification(ITU-TRecommendationE.164)andbeprecededbya“+”.Spacesandhyphensmaybeusedtosplitupaphonefieldtoaidreadabilityifdesired. [email protected]=+1617555-6011 ConnectionData(c=)connectioninformation—notrequiredifincludedinallmediainwhichmediaspecificconneciondataoverrideoverallsessionconnectiondata c= c=INIP4172.31.90.251 Ifthesessionismulticast,theconnectionaddresswillbeanIPmulticastgroupaddress.TTLshoudlbepresentinIPv4multicastaddress.IfconnectionisunicasttheaddresscontainstheunicastIPaddressoftheexpecteddatasourceordatarelayordatasink. Bandwidth(b=)interpretedaskilobitspersecondbydefault b=: EncryptionKeys(k=)OnlyisSDPisexchangedinsecureandtrustedchannel,keysvabeexcahngedonthisSDPfield.Althoughthisprocessisnotrecomended, k=clear:k=base64:k=uri:k=prompt Attributes(a=) extendstheSDPwithvalueslikeflags a=inactive,a=sendonly,a=sendrecv,a=recvonly MappingtheEncoderSpecfrom a=rtpmap:/[/] a=rtpmap:96opus/48000/2a=rtpmap:0PCMU/8000a=rtpmap:8PCMA/8000a=rtpmap:9G722/8000a=rtpmap:101telephone-event/48000a=rtpmap:97telephone-event/8000 ConferenecTypelike“broadcast”,“meeting”,“moderated”,“test”, a=type: Orientationportraitorlandscapeforwhiteboardsession a=orient: ICEcandidates a=ice-pwd:86701d63e2d96ec42268679a a=ice-ufrag:948a1316 a=rtcp-12133xr:rcvr-rtt=all:10000stat-summary=loss,dup,jitt,TTLvoip-metrics Framepersecondforvideo a=framerate: Qualitybetween0–10(10beststillimage,5default,0wrst) a=quality: FormatspecificParameters a=fmtp: a=rtpmap:114AMR-WB/16000/1 a=fmtp:114mode-change-capability=2;max-red=220 a=rtpmap:113AMR-WB/16000/1 a=fmtp:113octet-align=1;mode-change-capability=2;max-red=220 a=rtpmap:102AMR/8000/1 a=fmtp:102mode-change-capability=2;max-red=220 a=rtpmap:115AMR/8000/1 a=fmtp:115octet-align=1;mode-change-capability=2;max-red=220 a=rtpmap:105telephone-event/16000 a=fmtp:1050-15 a=rtpmap:101telephone-event/8000 a=fmtp:1010-15 TimeDescriptioninSDP Timing(t=)timethesessionisactive) t= Iftheissettozero,thenthesessionisnotbounded,thoughitwillnotbecomeactiveuntilafterthe.Iftheisalsozero,thesessionisregardedaspermanent. t=00 RepeatTimes(r=) zeroormorerepeattimesforschedulingasession r= timezoneadjustments(z=) z=…. usefulforscejdulingsessionduringtransationtodaylightvsavingtostandardtimeandviceversa MediaDescriptioninSDP ForRTP,thedefaultisthatonlytheeven-numberedportsareusedfordatawiththecorrespondingone-higheroddportsusedfortheRTCPbelongingtotheRTPsession m=… m=audio20098RTP/AVP0101 willstreamRTPon20098andRTCPon20099 FormultipletransportportspairsofRTP,RTCPstreamarespecified m=/… m=audio20098/2RTP/AVP0101willstreamonepaironRTP20098,RTCP20099andRTP20100,RTCP20101 Ifnon-contiguousportsarerequired,theymustbesignalledusingaseparateattributelikeexample,“a=rtcp:” AdditioanSDPfeatures:Inadditiontonormalunicastsessions,SDPcanalsoconverymulticastgroupaddressformediaonIPmulticastsession.Private(encryptionofSDP)orpublicsessionarenottreateddifferentlybySDPandtheyareentorelyafunctionofimplementingmechanismlikeSIPorSAP.OptiopnalSDPparamsincludeURI,Categorisation“a=cat:”,Internationalisationetc Example1:TypicalAudiocallSIPINVITEshowingSIPheadersinblueandSDPingreenbelow INVITEnbspsip:[email protected]/2.0 Via:SIP/2.0/UDPx.x.x.x:5060branch=z9hG4bK400fc6e6 From:"123456789"ltsip:[email protected]=as42e2ecf6 To:ltsip:[email protected] Contact:ltsip:[email protected] Call-ID:[email protected] CSeq:102INVITE User-Agent:nbspMatrixSwitch Date:Thu,22Dec200518:38:28GMT Allow:INVITE,ACK,CANCEL,OPTIONS,BYE,REFER Content-Type:application/sdp Content-Length:268 v=0 o=root1404014040INIP4x.x.x.x s=session c=INIP4x.x.x.x t=00 m=audio26784RTP/AVP0818101 a=rtpmap:0PCMU/8000 a=rtpmap:8PCMA/8000 a=rtpmap:18G729/8000 a=rtpmap:101telephone-event/8000 a=fmtp:1010-16 a=fmtp:18nbspannexb=no---- c=*(connectioninformation-optionalifincludedatsession-level) b=*(bandwidthinformation) a=*(zeroormoremediaattributelines) TheaboveSDPshows4supportedmediacodecsonaudiostreamwhichare0PCMU,8PCMA,18G729andfinally101usedfortelephoneevents.ItalsoshowsRTP/AVPasRTPprofileanddoesnotcontainanym=cideolinewhichshowsthatthisendpointdoesnotwantavideocall,onlyanaudioone. Example2:VideoVallSIPinvitefromLinphone SIPURIParams InternetAssignedNumberAuthority(IANA)UniversalResourceIdentifier(URI)ParameterRegistrydefinesURIparamsthatcanbesuedalongwithSIPscheme sip:user:password@host:port;uri-parameters?headers compparam signallingcompressionofSIPmessages sip:[email protected];comp=sigcompVia:SIP/2.0/UDPserver1.foo.com:5060;branch=z9hG4bK87a7;comp=sigcomp TheaobveexmapleindicatesthattherequesthastobecompressedusingSigComp transport-param SIPcanuseanynetworktransportprotocol.ParameternamesaredefinedforUDP(RFC768),TCP(RFC761),andSCTP(RFC2960).ForaSIPSURI,thetransportparameterMUSTindicateareliabletransport. “transport=” (“udp” /“tcp” /“sctp” /“tls” /“ws”/other-transport) sip:alice:[email protected];transport=tcp maddrpaarm Theserveraddress(detsiantionaddress,port,transport)tobecontactedforthisuser,overridinganyaddressderivedfromthehostfield. Althoughdiscouraged,maddrURIparamhasbeenusedasasimpleformofloosesourcerouting.ItallowsaURItospecifyaproxythatmustbetraverseden-routetothedestination. user-param “user=” (“phone” “ip” “dialstring” other-user) sip:1-212-555-1212:[email protected];user=phone sip:123;[email protected];user=dialstring method-param “method=”Method sip:atlanta.com;method=REGISTER?to=alice%40atlanta.com annc-parameters(announcement) ANNC-URLsip‑ind annc‑ind “@” hostport annc‑parameters uri‑parameters sip:[email protected];\;play=file://fs.example.net//clips/my-intro.dvi;\;content-type=video/mpeg%3bencode%d3314M-25/625-50 sip-ind -“sip:” /“sips:” annc-ind -“annc” annc-parameters“;” play‑param[“;” delay‑param][“;” duration‑param][“;” repeat‑param][“;” locale‑param][“;” variable‑params][“;” extension‑params] play-param–“play=” prompt‑url prompt-url–“/provisioned/” announcement‑id announcement-id =1*(ALPHA /DIGIT) content-param“content‑type=” MIME‑type VoiceXMLMediaServices dialog-param“voicexml=” vxml-url; vxml-urlfollowstheURIsyntax method-param–“method=” (“get” /“post”) postbody-param-“postbody=” token ccxml-param–“ccxml=” json‑value aai-param-“aai=” json‑value json-value–false /null /true /object /array /number /string sip:[email protected];\voicexml=http://appserver.example.com/promptcollect.vxml;\maxage=3600;maxstale=0 dialog-params(promptandcollect) DIALOG-URL =sip-ind dialog-ind “@” hostport dialog‑parameters ttl-param(time-to-live) ttlparameterdeterminesthetime-to-livevalueoftheUDPmulticastpacketandMUSTonlybeusedifmaddrisamulticastaddressandthetransportprotocolisUDP. sip:[email protected];maddr=239.255.255.1;ttl=15 causeparam “cause”EQUALStatus-Code;404Unknown/Notavailable;486Userbusy;408Noreply;302Unconditional;487Deflectionduringalerting;480Deflectionimmediateresponse;503Mobilesubscribernotreachable;380Servicenumbertranslation RFC8119–Section2 sip:[email protected];target=bob%40example.com;cause=486 SIPResponses 1xx—ProvisionalResponses responsethattellstoitsrecipientthattheassociatedrequestwasreceivedbutresultoftheprocessingisnotknownyetwhichcouldbeiftheprocessinghasntfinishedimmediately.Thesendermuststopretransmittingtherequestuponreceptionofaprovisionalresponse. 100Trying180Ringing:Triigersalocalringingatcallersdevice181CallisBeingForwarded:UsedbeforetraneferingtoanotherUAsuchasduringforkingortranfertovoicemailServer 182Queued 183SessioninProgress:conveysinformation.HeadersfieldorSDPbodyhasmordetailsaboutthecall.UsedinannouncementsandIVR+DTMFtoobybeingfollowedby“Earlymedia”. 199EarlyDialogTerminated 2xx—SuccessfulResponses finalresponsesexpressresultoftheprocessingoftheassociatedrequestandtheyterminatethetransactions. 200OK202Accepted204NoNotification 3xx—RedirectionResponses Redirectionresponsegivesinformationabouttheuser’snewlocationoranalternativeservicethatthecallershouldtryforthecall.Usedforcaseswhentheservercantsatisfythecallandwantsthecallertotryelsewhere.Afterthisthecallerissupposetoresendtherequesttothenewlocation. 300MultipleChoices301MovedPermanently302MovedTemporarily305UseProxy380AlternativeService 4xx—ClientFailureResponses negativefinalresponsesindicatingthattherequestcouldn’tbeprocessed duetocallersfault,forreasonssuchastcontainsbadsyntaxorcannotbefulfilledatthatserver. 400BadRequest401Unauthorized402PaymentRequired403Forbidden404NotFound405MethodNotAllowed406NotAcceptable407ProxyAuthenticationRequired408RequestTimeout409Conflict410Gone411LengthRequired412ConditionalRequestFailed413RequestEntityTooLarge414Request-URITooLong415UnsupportedMediaType416UnsupportedURIScheme417UnknownResource-Priority420BadExtension421ExtensionRequired422SessionIntervalTooSmall423IntervalTooBrief424BadLocationInformation428UseIdentityHeader429ProvideReferrerIdentity430FlowFailed433AnonymityDisallowed436BadIdentity-Info437UnsupportedCertificate438InvalidIdentityHeader439FirstHopLacksOutboundSupport470ConsentNeeded480TemporarilyUnavailable481Call/TransactionDoesNotExist482LoopDetected.483TooManyHops484AddressIncomplete485Ambiguous486BusyHere487RequestTerminated488NotAcceptableHere489BadEvent491RequestPending493Undecipherable494SecurityAgreementRequired 5xx—ServerFailureResponses negativeresponsesbutindicatingthatfaultisatserver’ssideforcasessuchasservercantordoesntwanttorespondthetherequest. 500ServerInternalError501NotImplemented502BadGateway503ServiceUnavailable504ServerTime-out505VersionNotSupported513MessageTooLarge580PreconditionFailure 6xx—GlobalFailureResponses requestcannotbefulfilledatanyserverwithdefinitiveinformation 600BusyEverywhere603Decline604DoesNotExistAnywhere606NotAcceptable MandatorySIPheadersinSIPrespone SIP/2.0200OK Via:SIP/2.0/UDPhost.domain.com:5060 From:Bob To:Altanai Call-ID:[email protected] CSeq:1INVITE Via,From,To,Call-ID,and  CSeq arecopiedexactlyfromrequest SIPVoIPsystem Architecture YoucanreadmoreaboutSIPbasedArchitecturehere:SIPbasedarchitecture Re-INVITEandTarget-RefreshRequestHandling AnINVITErequestsentwithinanexistingdialogisknownasare-INVITE.Are-Invitehasanoffer-answerexchangeandcanbeusedtodothefollowing changethesessionand/ordialogparamschangetheporttowhichmediashouldbesent.changetheconnectionaddressormediatype.Hold/ReleaseandSUSPEND/RESUMErtpstreams(connectionaddressiszero).FAX(T.38andBypass). Re-INVITEwithSDPuseCases 1.UASrejectsallchangesinparamsinre-INVITE SitutaionwhereUACestablishesaudioonlycall SDP1:m=audio30000RTP/AVP0 butlaterwantstoupgradetovideoaswellSDP: m=audio30000RTP/AVP0 m=video30002RTP/AVP31 UASconfiguredtorejectvideostreams,canrejectthiswitha4XXerrorandgetACK.Nochangestosessionaremade 2.UASreceivesre-INVITEforparambutwantstoacceptfewandrejectothers,itsendsbackSDPwithacceptablechangeswith200OK ForinstanceUACmovestohighbandwidthaccesspointandwantstoupdateIPofmediastream.Italsowansttoaddvideostream initialSDP m=audio30000RTP/AVP0c=INIP4192.0.2.1 newSDPinreINVITE m=audio30000RTP/AVP0c=INIP4192.0.2.2m=video30002RTP/AVP31c=INIP4192.0.2.2 UASreturnsa200(OK)responsetoacceptIPbutsetstheportofthevideostreamtozeroinitsSDPtoshowrejectedofvideostream. m=audio31000RTP/AVP0 c=INIP4192.0.2.5 m=video0RTP/AVP31 anotherexampleiswhenUACwwantstoaddanotheraudiocodecandalsoaddvideostreamtosession orignalSDP m=audio30000RTP/AVP0 c=INIP4192.0.2.1 re-inviteSDP m=audio30000RTP/AVP03c=INIP4192.0.2.1m=video30002RTP/AVP31c=INIP4192.0.2.1 againtheUASwilloptionallyacceptthesomeparamcangeslikeaudiocodebutsetvideotonullIPaddress m=audio31000RTP/AVP03 c=INIP4192.0.2.5 m=video31002RTP/AVP31 c=INIP40.0.0.0 3.UASreceivesre-INVITEbutwaitsforuserintervention UASreceivesre-INVITEtoaddvideo,butinsteadofrejecting,itpromptsusertopermit. SoUASprovidesanullIPaddressinsteadofsettingthestreamto‘inactive’becauseinactivestreamsstillneedtoexchangeRTPControlProtocol(RTCP)traffic m=audio31000RTP/AVP0c=INIP4192.0.2.5m=video31002RTP/AVP31c=INIP40.0.0.0 Laterifuserrejectstheadditionofthevideostream.Consequently,theUASsendsanUPDATErequest(6)settingtheportofthevideostreamtozeroinitsoffer. m=audio31000RTP/AVP0c=INIP4192.0.2.5m=video0RTP/AVP31c=INIP40.0.0.0 SipDetailed,Callflows,Architecturedescriptions,SIPservices,sipsecurity,sipprogrammingfromALTANAIBISHT References: RFC3261SIPRFC4566SDPhttps://tools.ietf.org/html/rfc4566RFC6141UpdatestheRFC3261withrespecttore-INVITEandUASbehaviourhttps://tools.ietf.org/html/rfc6141Techinvote:https://www.tech-invite.com/fo-abnf/tinv-fo-abnf-sipuriup.html SIPVoIPsystemarchitecture basicsJuly13,2013In"SIP"SIP(SessionInitiationProtocol )July13,2013In"SIP"HostedIP-PBXand SBCMarch5,2019In"TelecomInfo"SessionBordercontrollerfor WebRTCAugust2,2016In"WebRTCSaaS"KamailioDNSand NATFebruary2,2019In"Kamailio"OpensipsJune6,2018In"Opensips" Searchfor: Categories AccessandPhysicalLayer(7) AuxiliaryTechnologiesforVoIP(7) MOSandCallQuality(1) NaturalLanguageProcessing(NLP)(1) InternetofThings(7) BluetoothLowEnergy(1) Raspberrypi(2) RFID(1) Robotics(3) MediaProcesses(15) codecs(1) gstreamer(1) LiveStreamingandBroadcasting(10) VideoAnalytics(1) WowzaMediaServer(2) Protocols(6) XMPP(2) SessionInitiationProtocol(SIP)Frameworks(40) JAINSLEE(3) RCS(3) SIP(8) SIPservers(25) asterisk(1) freeswitch(5) Kamailio(11) Opensips(3) OracleSIPserver(1) SIPServlets(1) Signals(2) TelecomArchitectures(34) cloudtelephony(2) DataPrivacyandSIPsecurityinVoiceoverIP(5) IPMultimediaSubsystem(9) Legacytelecom(3) ServiceBroker(4) SIPmonitoringandNotification(1) TelecomInfo(8) VPN(1) WebRealTimeComm.(WEBRTC)(36) AugmentedReality(2) STUNandTURN(2) TangoFX(OpenSourceConferencingServer)-Archived(3) WebRTCMediaStack(6) WebRTCSaaS(5) WebRTCsecurity(2) WebRTCstandards(10) WebRTCusercasesandservice(6) Join568otherfollowers EmailAddress: Follow WebRTCIntegrator’sGuideTopPosts&Pages VOIPCallMetricMonitoringandMOS(MeanOpinionScore) SIPandSDPMessagesExplained KamailioWebRTCSIPServer SessionInitiationProtocol(SIP)ServiceCreation… GStreamer-1.8.1rtspserverandclientonubuntu RTPengineonkamailioSIPserver sipP(SIPtestingtool) WebRTCAudio/VideoCodecs KamailioasInbound/OutboundproxyorSessionBorderController(SBC) IMSinEPC(EvolvedPacketCore) altanai SpecializedinCPaaS,carrier-gradeWebRTC-SIPtelecomplatformsforUnifiedcommunication-collaboration,signalinggateways,SBC,softturrets,IoT-surveillanceandtelecomintegrations. ArdentcontributortoOpenSourcesoftware,avidfreelancer,innovatorandtechnicalwriter(telecom.altanai.com). Inventorof"RamuDroid"anIOTRoad-Cleaningrobot Authorofbook"WebRTCIntegrator'sGuide"publishedbyPackt PersonalLinks TelecomResearchandDevelopment ComputerScienceandEngineering RenewableEnergyTechbyAltanai CV github Ramudroid GoogleScholar VerifiedServices ViewFullProfile→ Tags2G 3G 4G Applicationprogramminginterface Applicationserver Arduino asterisk beaweblogic brekeke calea Callrouting Communicationsprotocol CSP eclipse FFMpeg freeswitch gsm GSMA H264 HTTPREST ICE identitymanagement IM IMS IN Instantmessaging IntelligentNetwork IOT IPaddress IPMultimediaSubsystem Jainslee Java JavaScript JSLEE Kamailio kapanga LTE MCU Mediaserver medistream NAT OTT pstn ramudroid raspberrypi RCS Real-timecommunication Real-timeTransportProtocol regulatoryconstrainswithwebrtc RTC RTCP RTMP RTP RTPengine SBC sdp Security Service-orientedarchitecture servicebroker Serviceharmonization SessionBorderController SessionInitiationProtocol Sip sipinvite sipserver socketio SRTP STUN Telecom TelecomEvolution Telecommunications TelecomServiceProvider TFX TURN unifiedcommunication VOIP WebRTC Wowza xlite XMPP CategoryCloudAccessandPhysicalLayerAugmentedRealityAuxiliaryTechnologiesforVoIPDataPrivacyandSIPsecurityinVoiceoverIPfreeswitchIPMultimediaSubsystemJAINSLEEKamailioLegacytelecomLiveStreamingandBroadcastingOpensipsProtocolsRaspberrypiRCSRoboticsServiceBrokerSignalsSIPSIPserversSTUNandTURNTangoFX(OpenSourceConferencingServer)-ArchivedTelecomArchitecturesTelecomInfoWebRTCMediaStackWebRTCSaaSWebRTCsecurityWebRTCstandardsWebRTCusercasesandserviceWowzaMediaServerXMPP June2022 M T W T F S S  12345 6789101112 13141516171819 20212223242526 27282930   «Mar     RecentComments BertHonEvolutionofvoice Commun…altanaionSIPVoIPsystemarchitecture…ReaderonSIPVoIPsystemarchitecture…VoIPdeveloperonEEP(formelyHEP)ExtensibleE…TaraEatononEvolutionofvoice Commun…PhilipStonesonUC(UnifiedCommunications)and…AnonymousonVideoCodecs–H264,H2… FollowmeonTwitterMyTweets RSSLinks RSS-Posts RSS-Comments TelecomR&DFaultToleranceandErrorCorrectioninWebRTCMarch10,2022FluctuatingNetworksDynamicBandwidthestimationJitterBufferDemandforHighQualityVideoTradeoffbetweenLatencyvsQualityBettercompressionalgorithmsvsCPUcomputeFullINTRA-frameRequest(FIR)PictureLossIndication(PLI)LayeringforadaptivestreamingRedundantEncoding(RED)inMediaPacketsCongestionFeedbackLoopReduceframequalityandr[…]AEC(EchoCancellation)andAGC(GainControl)inWebRTCMarch10,2022AcousticEchoHybrid/ElectronicEchoinPSTNphonesNoiseSuppressioninWebRTCEchoCancellationWebRTCEchoCancellationAutomaticGainControl(AGC)Echoisthesoundofyourownvoicereverberating.Iftheamplitudeofsuchasoundishighandintervalsexceed25ms,itbecomesdisruptivetotheconversation.Itstypescanbe…ContinuereadingAEC(Ec[…]VoIPAPIdesignDecember22,2021PublicAPIendpointsInternalAPIgatewaysAPIRateLimiterTokenbasedRateLimitingTokenbucketfilterHierarchicalTokenBucket(HTB)FairQueingCBQ(ClassBasedQueing)ModularQoScommand-Lineinterface(MQC)ShapingThrottlingVoIPmanagesCallsetupandteardownusingIPprotocol.TheAPIscanbeusedtoprovidepublicorinternalendpoinsttocreat[…]HighavailiabilityandScalibilityinVoIPplatformsDecember22,2021LoadBalancersMPLSService-discoveryKeepalive,unregisteringunhealthynodesReplicationDataStoreReplicationQuickResponse/LowlatencyScalabilityautoscallingPartitioiningMultiplePoPs(pointofpresence)MinimalLatencyandlowestamountoftarfficviapublicinternetHighavailiability(HA)59’sinaggregatefailuresHAforLoadbalancer(LB)H[…]EEP(formelyHEP)ExtensibleEncapsulationProtocolwithHOMERSeptember19,2021EEPduplicatesandIPdatagramandencapsulatesandsendsforremoterelatimemonitoringforSIPspecificalertsandnotifications.HEPispopularamongmanySIPserversincludingFreeswitch,Opensips,kamailio,RTPengineasanexternalmodule.intendedforpassiveduplicatedforremotecollectioncanbeusedforauditstorageandanalysisdoesnotalter[…]EnergyEfficientVoIPsystemsJuly19,2021DataCentresaretheconcentratedprocessingunitsfortheamazingInternetthatisdrivingthetechnologicalinnovationofourgenerationandhasbecomethebackboneofourglobaleconomy.DataCentresnotonlyprocess,storeandcarrytextualdataratheravastamountofcomputingisformultimediacontentwhichcouldrangefromsocialmedia…Continuerea[…]WhyLuaisagoodchoiceforScriptingcallconfigurationsinSIPserverslikeKamailioandFreeswitchDecember1,2020PrograminginSIPserversenablestheIPtelephonyprovidertoaddcomplexcontrolthatisdifficulttorealisewithsimpledialplanXMLandIVRmenus.ThesearebesthandledbyusingaprogramthatiscompiledwiththetelecomapplicationserverandinvokedbySIPrequestsorresponsesinthesession.Thismayincludeusing…ContinuereadingWhyLuaisag[…]TeleMedicineandWebRTCNovember24,2020Thesolutionenablesdoctors/nurses/medicalpractitionersandpatients todoHighdefinitionAudio/videocalls Endtoendencryptedp2pchats IntegrationwithHMS(hospitalmanagementsystem)tofetchhistoryofthepatients Screenssharingtoshowreportswithouttransferringthemasfiles IncludemoreconcernedpeopleofdoctorsusingMeshbasedpee[…] 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