Signaling Issues for Voice over MPLS in C#

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Signaling Issues for Voice over MPLS
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Figure 7-11 SIP components
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1 User agent client (UAC) An entity that initiates a call 2 User agent server (UAS) An entity that receives a call Both UAC and UAS can terminate a call The proxy server is an intermediary program that acts as both a server and a client to make requests on behalf of other clients Requests are serviced internally or by passing them on, possibly after translation, to other servers The server interprets, rewrites, or translates a Request message before forwarding it The location server is utilized by an SIP redirect or proxy server to obtain information about a called party s possible location(s) The redirect server is a server that accepts an SIP request, maps the address into zero or more new addresses, and returns these addresses to the client Unlike a proxy server, the redirect server does not initiate its own SIP request Unlike a UAS, the redirect server does not accept or terminate calls1
Figure 7-11 identifies key SIP components, which are described next A user agent is an application that initiates, receives, and terminates calls There are two types:
AM FL Y
SIP Components
Seven
The registrar server is a server that accepts register requests The registrar server may support authentication A registrar server is typically colocated with a proxy or redirect server and may offer location services SIP components communicate by exchanging SIP messages, as depicted in the following list1 SIP borrows much of the syntax and semantics from HTTP An SIP message looks like an HTTP message it has message formatting, a header, and MIME support The SIP address is identified by an SIP URL; the format of the URL is user@host
SIP Methods
SIP Responses
I I I I I I
INVITE Initiates a call by inviting the user to participate in the session ACK Confirms that the client has received a final response to an Invite request BYE Indicates a termination of the call CANCEL Cancels a pending request REGISTER Registers the user agent OPTIONS Used to query the capabilities of a server INFO Used to carry out-ofbound information, such as Dual Tone Multiple Frequency (DTMF) digits
1xx Informational messages 2xx Successful responses 3xx Redirection responses 4xx Request failure responses 5xx Server failure responses 6xx Global failure responses
I I I I I
Establishing communication using SIP usually takes place in six steps (see Figure 7-12):1 1 Register, initiate, and locate the user 2 Determine which media to use involves delivering a description of the session to which the user is invited 3 Determine the willingness of the called party to communicate The called party must send a Response message to indicate a willingness to communicate: accept or reject
Signaling Issues for Voice over MPLS
Figure 7-12 SIP call handling
IP User Agent INVITE Proxy Server Location/Redirect Server INVITE 302 (Moved Temporarily) ACK INVITE Call Setup INVITE 302 (Moved Temporarily) ACK INVITE 180 (Ringing) 200 (OK) ACK 180 (Ringing) 200 (OK) ACK 180 (Ringing) 200 (OK) ACK Proxy Server IP User Agent
Media Path Call Teardown BYE 200 (OK)
RTP MEDIA PATH BYE 200 (OK) BYE 200 (OK)
4 Set up the call 5 Modify or handle the call 6 Terminate the call SIP was designed for integration with IETF environments Existing IETF protocol standards can be used to build an SIP application The protocol works with existing IETF protocols such as the Resource Reservation Protocol (RSVP) (reserve network resources), RTP (transport real-time data and provide QoS feedback), the Real-Time Streaming Protocol (RTSP) (controls the delivery of streaming media), the Session Advertisement Protocol (SAP) (advertising multimedia session via multicast), the Session Description Protocol (SDP) (describing multimedia sessions), MIME (content description), HTTP (web pages delivery), and other IETF protocols SIP supports flexible and intuitive feature creation using SIP-Common Gateway Interface (SIP-CGI) and Call Processing Language (CPL)
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