Examples Of SIP Message Sequences - 國立臺灣大學
Examples of SIP Message Sequences Via:From: and To:Call-ID: Contact: (for future SIPmessage transmission) Zero, no msg bodyCSeq: *Content-Length: host-specificA response to any requestmust use the same value ofCSeq as used in the request.Expires: TTL0, unregIP Telephony1
Invitation A two-party call Subject: Content-Type: optionalapplication/sdpA dialog ID To identify a peer-to-peerrelationship between twouser agentsTag in FromTag in ToCall-ID
Termination of a Call CSeq has changed.Daniel sip:Collins@work.com Boss sip:Manager@station2.work.com aBYE sip:manager@work.com SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch z9hG4bK123Max-Forwards: 70From: Daniel sip:Collins@work.com ; tag 44551To: Boss sip:Manager@station2.work.com ; tag 11222Call-ID: 123456@station1.work.comCSeq: 2 BYEContent-Length: 0bSIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com;branch z9hG4bK123From: Daniel sip:Collins@work.com ; tag 44551To: Boss sip:Manager@station2.work.com ; tag 11222Call-ID: 123456@station1.work.comCSeq: 2 BYEContent-Length: 0IP Telephony3
Redirect Servers An alternative address 302, Moved temporarilyAnother INVITE Same Call-IDCSeq
Proxy Servers [1/2] Sits between a user-agent client and the far-end useragent serverNumerous proxies can reside in a chain between thecaller and callee. The most common scenario will have at least two proxies: oneat the caller and one at the callee end.It is likely that only the last proxy in the chain changes theRequest-URI.The other proxies in the chain would simply use the domainpart of the received Request-URI as input to a locationfunction (e.g., DNS) to determine the next hop.IP Telephony5
Proxy Servers [2/2] Via: The path taken by a requestLoop detected, 482 (status code)For a response The first Via: header is checked and removed.The second Via: header is checked. If it exists, perform forwarding.If not, the response is destined to the proxy itself.The response finds its way back to the originator of the request.Branch: used to distinguish between multiple responses to thesame request Forking Proxy: Issue a single request to multiple destinationsIP Telephony6
Proxy State [1/2] Can be either stateless or statefulIf stateless, the proxy takes an incoming request,performs whatever translation and forwards thecorresponding outgoing request and forgetsanything. Retransmission takes the same path (no change onretransmission).If stateful, the proxy remembers incoming requestsand corresponding outgoing request. The proxy is able to act more intelligently on subsequentrequests and responses related to the same session.IP Telephony7
Proxy State [2/2] Record-Route: and Route: Headers The subsequent requests may not pass through the samepath as the initial request/response. A Proxy might require that it remains in the signaling pathfor all subsequent requests to provide some advancedservice. E.g., use Contact:In particular for a stateful proxyInsert its address into the Record-Route: headerThe response includes the Record-Route: headerThe information contained in the Record-Route: header isused in the subsequent requests related to the same call.The Route: header is used to record the path that therequest is enforced to pass.IP Telephony8
Forking Proxy A proxy can “fork” requestsA user is registered at several locations ;branch xxxIn order to handle such forking, a proxy must bestateful.IP Telephony11
The Session Description Protocol The Most Common Message Body Session information describing the media to beexchanged between the partiesSDP, RFC 2327 (initial publication) A number of modifications to the protocol have beensuggested.SIP uses SDP in an answer/offer mode. An agreement between the two parties as to thetypes of media they are willing to shareRFC 3264 (An Offer/Answer Model with SDP) To describe how SDP and SIP should be used togetherIP Telephony14
The Structure of SDP SDP simply provides a format for describingsession information to potential sessionparticipants.Text-based ProtocolSession DescriptionSession Level InformationThe Structure of SDP Session Level Info Name of the sessionOriginator of the sessionTime that the session is to be activeMedia Level Info Media typePort numberTransport protocolMedia formatProtocol VersionOriginator and Session IDSession NameSession TimeMedia Description 1Media Name and TransportConnection InformationMedia Description 2Media Name and TransportConnection InformationIP Telephony15
SDP Syntax A number of lines of textIn each line field valuefield is exactly one character (case-significant)Session-level fieldsMedia-level fields Begin with media description field (m )IP Telephony16
Mandatory Fields v (protocol version)o (session origin or creator)s (session name), a text string t (time of the session), the start time and stop time For multicast conferenceFor pre-arranged multicast conferencem (media) Media typeThe transport portThe transport protocolThe media format (typically an RTP payload format)IP Telephony17
Optional Fields [1/3] Some optional fields can be applied at bothsession and media levels. i (session information) The value applied at the media level overrides that at thesession levelA text descriptionAt both session and media levelsIt would be somewhat superfluous since SIP alreadysupports the Subject header.u (URI of description) Where further session information can be obtainedOnly at session levelIP Telephony18
Optional Fields [2/3] e (e-mail address) p (phone number) Only at the session levelc (connection information) Who is responsible for the sessionOnly at the session levelNetwork type, address type and connection addressAt session or media levelb (bandwidth information) In kilobits per secondAt session or media levelIP Telephony19
Optional Fields [3/3] r (repeat times) z (timezone adjustments) For regularly scheduled sessionStandard time and daylight savings timek (encryption key) For regularly scheduled session a session is to be repeatedHow often and how many timesAn encryption key or a mechanism to obtain it for thepurposes of encrypting and decrypting the mediaAt session or media levela (attributes) Describe additional attributesIP Telephony20
Ordering of Fields Session Level Protocol version (v)Origin (o)Session name (s)Session information (i)URI (u)E-mail address (e)Phone number (p)Connection info (c)Bandwidth info (b)Time description (t)Repeat info (r)Time zone adjustments (z)Encryption key (k)Attributes (a) Media level Media description (m)Media info (i)Connection info (c) Optional if specified at thesession levelBandwidth info (b)Encryption key (k)Attributes (a)IP Telephony21
Subfields [1/3] Field value of subfield1 value of subfield2 value of subfield3 Origin Username, the originator’s login id or “-”Session ID Version, a version number for this particular sessionNetwork type A text stringIN refers to InternetAddress type A unique IDMake use of NTP timestampIP4, IP6Address, a fully-qualified domain name or the IP addressIP Telephony22
Subfields [2/3] Connection Data The network and address at which media data will bereceivedNetwork typeAddress typeConnection addressMedia Information Media type PortFormat Audio, video, data, or controlList the various types of media format that can be supportedAccording to the RTP audio/video profilem audio 45678 RTP/AVP 15 3 0 G.728, GSM, G.711IP Telephony23
Subfields [3/3] Attributes To enable additional information to be includedProperty attribute Value attribute a sendonlya recvonlya orient:landscape used in a shared whiteboard sessionRtpmap attribute The use of dynamic payload typea rtpmap: payload type encoding name / clock rate [/ encoding parameters ].m video 54678 RTP/AVP 98a rtpmap 98 L16/16000/2 16-bit linear encoded stereo (2 channels) audio sampledat 16kHzIP Telephony24
Usage of SDP with SIP SIP and SDP make a wonderful partnershipfor the transmission of session information.SIP provides the messaging mechanism forthe establishment of multimedia sessions.SDP provides a structured language fordescribing the sessions. The entity headers identifies the message body.IP Telephony25
SIP Inclusion in SIP Messages Fig 5-15 INVITE with multiple media streams G.728 is selectedUnsupported should also be returned with a port number ofzeroAn alternative INVITEm audio 4444 RTP/AVP 2 4 15a rtpmap 2 G726-32/8000a rtpmap 4 G723/8000a rtpmap 15 G728/8000 200 OKm audio 6666 RTP/AVP 15a rtpmap 15 G728/8000IP Telephony26
Daniel sip:Collins@station1.work.com abBoss sip:Manager@station2.work.com INVITE sip:Manager@station2.work.com SIP/2.0From: Daniel sip:Collins@station1.work.com ; tag abcd1234To: Boss sip:Manager@station2.work.com CSeq: 1 INVITEContent-Length: 213Content-Type: application/sdpContent-Disposition: sessionv 0o collins 123456 001 IN IP4 station1.work.coms c IN IP4 station1.work.comt 0 0m audio 4444 RTP/AVP 2a rtpmap 2 G726-32/8000m audio 4666 RTP/AVP 4a rtpmap 4 G723/8000m audio 4888 RTP/AVP 15a rtpmap 15 G728/8000SIP/2.0 200 OK IP Telephony27
Boss sip:Manager@station2.work.com Daniel sip:Collins@station1.work.com bcdSIP/2.0 200 OKFrom: Daniel sip:Collins@station1.work.com ; tag abcd1234To: Boss sip:Manager@station2.work.com ; tag xyz789CSeq: 1 INVITEContent-Length: 163Content-Type: application/sdpContent-Disposition: sessionv 0o collins 45678 001 IN IP4 station2.work.coms c IN IP4 station2.work.comt 0 0m audio 0 RTP/AVP 2m audio 0 RTP/AVP 4m audio 6666 RTP/AVP 15a rtpmap 15 G728/8000ACK sip:manager@station2.work.com SIP/2.0From: Daniel sip:Collins@station1.work.com ; tag abcd1234To: Boss sip:Manager@station2.work.com ; tag xyz789CSeq: 1 ACKContent-Length: 0ConversationIP Telephony28
SIP and SDP Offer/Answer Model Re-INVITE is issued when the server replies with morethan one codec. With the same dialog identifier (To and From headers, includingtag values), Call-ID and Request-URIThe session version is increased by 1 in o line of message body.A mismatch 488 or 606Not AcceptableA Warning header with warning code 304 (media type notavailable) or 305 (incompatible media type)Then the caller issues a new INVITE request.IP Telephony29
Daniel sip:Collins@station1.work.com abBoss sip:Manager@station2.work.com INVITE sip:manager@station2.work.com SIP/2.0CSeq: 1 INVITEContent-Length: 183Content-Type: application/sdpContent-Disposition: sessionv 0o collins 123456 001 IN IP4 station1.work.coms c IN IP4 station1.work.comt 0 0m audio 4444 RTP/AVP 2 4 15a rtpmap 2 G726-32/8000a rtpmap 4 G723/8000a rtpmap 15 G728/8000a inactiveSIP/2.0 200 OKCSeq: 1 INVITEContent-Length: 157Content-Type: application/sdpContent-Disposition: sessionv 0o collins 45678 001 IN IP4 station2.work.coms c IN IP4 station2.work.comt 0 0m audio 6666 RTP/AVP 4 15a rtpmap 4 G723/8000a rtpmap 15 G728/8000a inactiveIP Telephony30
Daniel sip:Collins@station1.work.com Boss sip:Manager@station2.work.com cACK sip:manager@station2.work.com SIP/2.0From: Daniel sip:Collins@station1.work.com ; tag abcd1234To: Boss sip:Manager@station2.work.com ; tag xyz789CSeq: 1 ACKContent-Length: 0dINVITE sip:manager@station2.work.com SIP/2.0CSeq: 2 INVITEContent-Length: 126Content-Type: application/sdpContent-Disposition: sessionv 0o collins 123456 002 IN IP4 station1.work.coms c IN IP4 station1.work.comt 0 0m audio 4444 RTP/AVP 15a rtpmap 15 G728/8000IP Telephony31
OPTIONS Method Determine the capabilities of a potential calledpartyAccept Header Allow Header Indicate the type of information that the sender hopes toreceiveIndicate the SIP methods that Boss can handleSupported Header Indicate the SIP extensions that can be supportedIP Telephony32
Daniel sip:Collins@station1.work.com abBoss sip:Manager@station2.work.com OPTIONS sip:manager@station2.work.com SIP/2.0Via: SIP/2.0/UDP Station1.work.com; branch z9hG4bK7890123From: Daniel sip:Collins@work.com ; tag lmnop123To: Boss sip:Manager@station2.work.com Call-ID: 123456@station1.work.comContact: Daniel sip:Collins@station1.work.com CSeq: 1 OPTIONSAccept: application/sdpContent-Length: 0SIP/2.0 200 OKVia: SIP/2.0/UDP Station1.work.com; branch z9hG4bK7890123From: Daniel sip:Collins@work.com ; tag lmnop123To: Boss sip:Manager@station2.work.com ; tag xyz5678Call-ID: 123456@station1.work.comCSeq: 1 OPTIONSAllow: INVITE, ACK, CANCEL, OPTIONS, BYESupported: newfieldContent-Length: 146Content-Type: application/sdpv 0o manager 45678 001 IN IP4 station2.work.coms c IN IP4 station2.work.comt 0 0m audio 0 RTP/AVP 4 15a rtpmap 4 G723/8000a rtpmap 15 G728/8000IP Telephony33
SIP Extensions and Enhancements RFC 2543, March 1999 RFC 3261, June 2002SIP has attracted enormous interest.Traditional telecommunications companies, cableTV providers and ISPA large number of extensions to SIP havebeen proposed. SIP will be enhanced considerably before itbecomes an Internet standard.IP Telephony34
183 Session Progress It has been included within the revised SIPspec. To open one-way audio path from called end tocalling end Enable in-band call progress information to be transmitted Tones or announcementsInterworking with SS7 network ACM (Address Complete Message)For SIP-PSTN-SIP connectionsIP Telephony35
The Supported Header The Base RFC 2543 The Require: Header In request (client - server) The Unsupported Header In response (server - client) A client indicates that a server must support certain extension.420 (bad extension)A cumbersome way of determining what extensions a serverdoes or does not supportThe Supported: Header (RFC 3261) May be included in OPTIONS request Associated with the Supported: header is 421 (extensionrequired) response.Can also be included in responsesIP Telephony36
SIP INFO Method Be specified in RFC 2976For transferring information during anongoing session DTMF digits, account-balance information, mid-callsignaling information (from PSTN)Application-layer information could be transferredin the middle of a call.A powerful, flexible tool to support newservicesIP Telephony37
SIP Event Notification Several SIP-basedapplications have beendevised based on theconcept of a user beinginformed of some event. E.g., Instant messagingRFC 3265 has addressedthe issue of eventnotification. SUBSCRIBE and NOTIFYThe Event headerSubscriberaNotifierSUBSCRIBEb200 OKcNOTIFYCurrent stateinformationNOTIFYUpdated stateinformationd 200 OKef200 OKIP Telephony38
SIP for Instant Messaging The IETF working group – SIP for InstantMessaging and Presence Leveraging Extensions(SIMPLE)A new SIP method – MESSAGE This request carries the actual message in amessage body.A MESSAGE request does not establish a SIP dialog.IP Telephony39
Boss sip:Manager@pc1.home.com sip:Server.work.comab MESSAGE sip:Collins@work.com SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch z9hG4bK7890Max-Forwards: 70From: Boss sip:Manager@home.net To: Daniel sip:Collins@work.com Call-ID: 123456@pc1.home.netCSeq: 1 MESSAGEContent-Type: text/plainContent-Length: 19Content-Disposition: renderHello. How are you?Daniel sip:Collins@station1.work.com MESSAGE sip:Collins@work.com SIP/2.0Via: SIP/2.0/UDP server.work.com; branch z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch z9hG4bK7890Max-Forwards: 69From: Boss sip:Manager@home.net To: Daniel sip:Collins@work.com Call-ID: 123456@pc1.home.netCSeq: 1 MESSAGEContent-Type: text/plainContent-Length: 19Content-Disposition: renderHello. How are you?cdSIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch z9hG4bKxyz1SIP/2.0 200 OKVia: SIP/2.0/UDP pc1.home.net; branch z9hG4bK7890 Via: SIP/2.0/UDP pc1.home.net; branch z9hG4bK7890From: Boss sip:Manager@home.net From: Boss sip:Manager@home.net To: Daniel sip:Collins@work.com To: Daniel sip:Collins@work.com Call-ID: 123456@pc1.home.netCall-ID: 123456@pc1.home.netCSeq: 1 MESSAGECSeq: 1 MESSAGEContent-Length: 0Content-Length: 0IP Telephony40
Boss sip:Manager@pc1.home.com sip:Server.work.comefMESSAGE sip:Manager@home.net SIP/2.0Via: SIP/2.0/UDP server.work.com; branch z9hG4bKabcdVia: SIP/2.0/UDP station1.work.com; branch z9hG4bK123Max-Forwards: 69From: Daniel sip:Collins@work.com To: Boss sip:Manager@home.net Call-ID: 456789@station1.work.comCSeq: 1101 MESSAGEContent-Type: text/plainContent-Length: 22Content-Disposition: renderDaniel sip:Collins@station1.work.com MESSAGE sip:Manager@home.net SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch z9hG4bK123Max-Forwards: 70From: Daniel sip:Collins@work.com To: Boss sip:Manager@home.net Call-ID: 456789@station1.work.comCSeq: 1101 MESSAGEContent-Type: text/plainContent-Length: 22Content-Disposition: renderI’m fine. How are you?I’m fine. How are you?gh SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch z9hG4bKabcdVia: SIP/2.0/UDP station1.work.com; branch z9hG4bK123From: Daniel sip:Collins@work.com To: Boss sip:Manager@home.net Call-ID: 456789@station1.work.comCSeq: 1101 MESSAGEContent-Length: 0SIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com; branch z9hG4bK123From: Daniel sip:Collins@work.com To: Boss sip:Manager@home.net Call-ID: 456789@station1.work.comCSeq: 1101 MESSAGEContent-Length: 0IP Telephony41
SIP REFER Method To enable the sender of the request to instruct thereceiver to contact a third party With the contact details for the third party included within the REFERrequestFor Call Transfer applicationsThe Refer-to: and Refer-by: HeadersThe dialog between Mary and Joe remains established. Joe could return to the dialog after consultation with Susan.IP Telephony42
omsip:Susan@station3.work.comREFER sip:Joe@station2.work.com SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch z9hG4bK789Max-Forwards: 70From: Mary sip:Mary@work.com ; tag 123456To: Joe sip:Joe@work.com ; tag 67890Contact: Mary Mary@station1.work.com Refer-To: Sussan sip:Sussan@station3.work.com Call-ID: 123456@station1.work.comCSeq: 123 REFERContent-Length: 0bSIP/2.0 202 AcceptedcVia: SIP/2.0/UDP station1.work.com; branch z9hG4bK789From: Mary sip:Mary@work.com ; tag 123456To: Joe sip:Joe@work.com ; tag 67890Contact: Joe Joe@station2.work.com Call-ID: 123456@station1.work.comCSeq: 123 REFERContent-Length: 0INVITE sip:Susan@station3.work.com SIP/2.0Via: SIP/2.0/UDP station2.work.com; branch z9hG4bKxyz1Max-Forwards: 70From: Joe sip:Joe@work.com ; tag abcxyzTo: Susan sip:Susan@station3.work.com Contact: Joe Joe@station2.work.com Call-ID: 67890@station2.work.comCSeq: 567 INVITEContent-Type: application/sdpContent-Length: xxContent-Disposition: session{message body}IP Telephony43
omsip:Susan@station3.work.comSIP/2.0 200 OKVia: SIP/2.0/UDP station2.work.com; branch z9hG4bKxyz1From: Joe sip:Joe@work.com ; tag abcxyzTo: Susan sip:Susan@station3.work.com ; tag 123xyzCall-ID: 67890@station2.work.comCSeq: 567 INVITEContent-Type: application/sdpContent-Length: xxContent-Disposition: session{message body}fACK sip:Susan@station3.work.com SIP/2.0gVia: SIP/2.0/UDP station2.work.com; branch z9hG4bKxyz1NOTIFY sip:Mary@station1.work.com SIP/2.0Max-Forwards: 70Via: SIP/2.0/UDP station2.work.com; branch z9hG4bK123 From: Joe sip:Joe@work.com ; tag abcxyzMax-Forwards: 70To: Susan sip:Susan@station3.work.com ; tag 123xyzFrom: Joe sip:Joe@work.com Call-ID: 67890@station2.work.comTo: Mary sip:Mary@work.com CSeq: 567 ACKContact: Joe Joe@station2.work.com Content-Length: 0Call-ID: 123456@station1.work.comCSeq: 124 NOTIFYContent-Type: message/sipfrag;version 2.0Content-Length: 15hSIP/2.0 200 OKVia: SIP/2.0/UDP station2.work.com; branch z9hG4bK123From: Joe sip:Joe@work.com To: Mary sip:Mary@work.com Call-ID: 123456@station1.work.comCSeq: 124 NOTIFYIP Telephony 44Content-Length: 0
Reliability of Provisional Responses [1/2] Provisional Responses If the messages is sent over UDP 100 (trying), 180 (ringing), 183 (session in progress)Are not answered with an ACKUnreliableLost provisional response may cause problems wheninteroperating with other network 180, 183 Q.931 alerting or ISUP ACMTo drive a state machineE.g., a call to an unassigned number ACM to create a one-way path to relay an announcement such as“The number you have called has been changed”If the provisional response is lost, the called might left in the darkand not understand why the call did not connect.IP Telephony45
Reliability of Provisional Responses [2/2] RFC 3262 Response ACKIn PRACKRSeq CSeqApply to 100Default timer value 0.5 sINVITE sip:ServerB@network.com SIP/2.0Via: SIP/2.0/UDP ClientA.network.com; branch z9hG4bK7890123Supported: 100relRequire: 100relFrom: sip:ClientA@network.com; tag lmnop123To: sip:ServerB@network.comCall-ID: 123456@ClientA.network.comCSeq: 1 INVITE?ResponseLostProv. Resp. ACKShould not bPRACK aResponse Seq 1, when retxmRAck Header ServerB@network.comReliability of ProvisionalResponses in SIPSupported: 100relRSeq Header ClientA@network.comcSIP/2.0 180 RingingVia: SIP/2.0/UDP ClientA.network.com; branch z9hG4bK7890123Require: 100relRSeq: 567890From: sip:ClientA@network.com; tag lmnop123To: sip:ServerB@network.com; tag xyz123Call-ID: 123456@ClientA.network.comCSeq: 1 INVITEResponseRetransmitSIP/2.0 180 RingingVia: SIP/2.0/UDP ClientA.network.com; branch z9hG4bK7890123Require: 100rel.IP Telephony46
IP Telephony47
The SIP UPDATE Method To enable the modification of sessioninformation before a final response to anINVITE is received The dialog is in the early state (An INVITE thatreceives a 183 response that includes a messagebody) The message body might establish a media stream fromcallee to caller for sending a ring tone or music while thecalled party is alerted.The UPDATE method can be used to change thecodecAnother important usage is when reservingnetwork resources as part of a SIP sessionestablishment.IP Telephony48
Integration of SIP Signaling and ResourceManagement [1/2] Ensuring that sufficient resources are available to handle amedia stream is very important. The signaling might take a different path from the media. To provide a high-quality service for a carrier-grade networkThe successful transfer of signaling messages does notimply to a successful transfer of media.Assume resource-reservation mechanisms are available(Chapter 8) On a per-session basis End-to-end network resources are reserved as part of sessionestablishment.On an aggregate basis A certain amount of network resources are reserved in advancefor a certain type of usage.Policing functions at the edge of the networkIP Telephony49
Integration of SIP Signaling and ResourceManagement [2/2]UserA@network.com Reserving networkresources in advance ofaltering the called userA new draft –“Integration of ResourceManagement and SIP” By using the provisionalresponses and UPDATEmethodBy involving extensions toSDPaUserB@network.comINVITESession Description(with pre-condition attributes)bcSIP/2.0 183 Session ProgressSession Description(with pre-condition attributes)PRACKdSIP/2.0 200(OK) (for PRACK)efResource ReservationUPDATESession Description(with updated pre-condition attributes)gSIP/2.0 200 (OK) (for UPDATE)Session Description(with updated pre-condition attributes)hiSIP/2.0 180 (Ringing) (response to initial INVITE)PRACK (for 180 response)jSIP/2.0 200(OK) (for PRACK)klSIP/2.0 200(OK) (for INVITE)ACKIP Telephony50
Example of e2e Resource Reservation [1/2] SDP for initial INVITE SDP for 183 responsev 0o userA 45678 001 IN IP4 stationA.network.coms c IN IP4 stationA.nework.comt 0 0m audio 4444 RTP/AVP 0a curr: qos e2e nonea des: qos mandatory e2e sendrecvv 0o userB 12345 001 IN IP4 stationB.network.coms c IN IP4 stationB.nework.comt 0 0m audio 6666 RTP/AVP 0a curr: qos e2e nonea des: qos mandatory e2e sendrecva conf: qos e2e recvIP Telephony51
Example of e2e Resource Reservation [2/2] SDP for UPDATE SDP for 200 responsev 0o userA 45678 001 IN IP4 stationA.network.coms c IN IP4 stationA.nework.comt 0 0m audio 4444 RTP/AVP 0a curr: qos e2e senda des: qos mandatory e2e sendrecvv 0o userB 12345 001 IN IP4 stationB.network.coms c IN IP4 stationB.nework.comt 0 0m audio 6666 RTP/AVP 0a curr: qos e2e sendrecva des: qos mandatory e2e sendrecvIP Telephony52
Example of Aggregatebased Reservation Each participant deals withnetwork access permission atits own end.Mandatory means that thesession can not continue unlessthe required resources aredefinitely available.None is the initial situation andindicates that no effort toreserve resources has yet takenplace.Response 580 (preconditionfailure)
Usage of SIP for Features/Services [1/2] Call-transfer application (with REFER method)Personal Mobility through the use of registrationOne number service through forking proxyCall-completion services by using Retry-After: headerTo carry MIME content as well as an SDP description SIP address is a URL To include a piece of text, an HTML document, an image and soonClick-to-call applicationsThe existing supplementary services in traditionaltelephony Call waiting, call forwarding, multi-party calling, call screeningIP Telephony54
Usage of SIP for Features/Services [2/2] Proxy invokes various types of advanced feature logic. Policy server (call-routing, QoS)Authentication serverUse the services of an IN SCP over INAPThe network might use the Parley Open Service Access(OSA) approach, utilizing application programminginterfaces (APIs) between the nodes.IP Telephony55
Call Forwarding On busy486, busy hereWith the same To, User 3can recognize that this callis a forwarded call,originally sent to User 2.Contact: header in 200responseCall-forwarding-on-noanswer Timeout CANCEL method
Consultation Hold A SIP UPDATEUser A asks User B aquestion, and User B needto check with User C forthe correct answer.If User C needs to talk toUser A directly, User Bcould use the REFERmethod to transfer thecall to User C.
PSTN Interworking PSTN Interworking Seamless interworkingbetween two differentprotocols is not quite easy. One-to-one mapping betweenthese protocolsPSTN – SIP – PSTN A SIP URL to a telephonenumberA network gatewayMIME media typesFor ISUPSIP for Telephony (SIP-T)The whole issue ofinterworking with SS7 isfundamental to the success ofVoIP in the real world.
Interworking with H.323 SIP-H.323 interworking gatewayIP Telephony59
IP Telephony60
Summary The future for signaling in VoIP networks Simple, yet flexibleEasier to implementFit well with the media gateway control protocols Coexisting with PSTNSIP is the protocol of choice for the evolutionof third-generation wireless networks. SIP-based mobile devices will become available.SIP-based network elements will be introducedwithin mobile networks.IP Telephony63
IP Telephony 5 Proxy Servers [1/2] Sits between a user-agent client and the far-end user- agent server Numerous proxies can reside in a chain between the caller and callee. The most common scenario will have at least two proxies: one at the caller and one at the callee end. It is likely that only the last proxy in the chain changes the Request-URI.
SIP SIP phones Blustar 8000i NA SIP SIP phones 9112i, 9133i, 480i Not Supported SIP SIP phones 673xi ( A673xi), 675xi ( A675xi) NA SIP SIP phones 6735i, 6737i ( A6735i, A6737i) NA SIP SIP phones 6739i NA SIP SIP phones 6863i, 6865i, 6867i NA SIP MiVoice Conference phone (UC360
C O N T E N T S Configuration of SIP Trunking for PSTN Access SIP-to-SIP 1 Finding Feature Information 1 Configuration of SIP Trunking for PSTN Access SIP-to-SIP Features 1 Configuring SIP Registration Proxy on Cisco UBE 3 Finding Feature Information 3 Registration Pass-Through Modes 4 End-to-End Mode 4 Peer-to-Peer Mode 5 Registration in Different Registrar Modes 7
How To Guide: SIP Trunking Configuration Using the SIP Trunk Page 6(19) 2.2 The SIP Trunk Page The SIP Trunk pages are found under SIP Trunks. Several SIP Trunk pages may be defined if you have several PBXs or Trunk Services. You need to purchase Additional Trunk Group licensees to get more than one SIP Trunk page. Details are found below. s d he n
To support SIP trunks through a SIP trunk service provider, the SIP Trunk Groups folder was added to the SIP Peers folder in DB Programming. To create a SIP Trunk Group for Fusion Connect Service Provider, navigate to System- Device and Feature Codes- SIP Peers- SIP Trunk Groups and right click in the right hand pane. Then select "Create SIP .
Call Flow Scenarios for Successful Calls This section describes call flows for the following scenarios, which illustrate successful calls: SIP Gateway-to-SIP Gateway—Call Setup and Disconnect, page 7-3 SIP Gateway-to-SIP Gateway—Call via SIP Redirect Server, page 7-6 SIP Gateway-to-SIP Gateway—Call via SIP Proxy Server, page 7-9
4. SIP, VVoIP and QoS 5. SIP and Media Security 6. STIR/SHAKEN and the 'identity' problem 7. Firewalls, NAT and Session Border Controllers 8. SIP Trunking 9. Testing, Troubleshooting and Interoperability 10. ENUM, Peering and Interconnect 11. SIP in the Cloud 12. SIP in Cellular networks 13. SIP and Fax over IP 14. SIP in UC, UCaaS and .
How to Guide: SIP Trunking Configuration using the SIP Trunks page 4 2.2 The SIP Trunk Page The SIP Trunk pages are found under SIP Trunks. Several SIP Trunk pages may be defined if you have several PBXs or Trunk Services. You need to purchase Additional Trunk Group licensees to get more than one SIP Trunk page. Details are found below. s d he Tru
STI-AS IBCF/ TrGW SIP UA Verifier 4. Get Private Key SKS 1. SIP INVITE 22. 200 OK 9. SIP INVITE IBCF/ TrGW CSCF STI-CR CVT 2. SIP INVITE 5. Private Key 7. SIP INVITE (with Identity) 8. SIP INVITE 10. SIP INVITE 11. SIP INVITE 13. Get Certificate 14. Certificate 16. Invoke Analytics 17. Result of Analytics 18. SIP INVITE (with Verification .
