SIP Messages

In SIP , you will find 2 types of SIP Messages, there are:

  • Request: A message sent by a client to a server that is used to invoke certains operations or functions.
  • Response : A message sent by a server to a client that indicates the status of the request received from the client.

Here are now the request messages which invoke function on the server ( note that these functions are also called methods)

  • Invite : When UAC want to initiate a session, it sends an Invite request to a server. When the request is received by the UAS, it process it and sends the appropriate response message.
  • Ack : This message is sent in reply to a final response message from a server.
  • Bye : Used to terminate a session
  • Cancel : a Cancel request is used to terminate a pending request ( a request for which a final response has not yet been received)
  • Register : This message is used to register contact information.
  • Options : An UA can query another UA or SIP Proxy server about its capabilities using Options request. In this way, a client can find out capabilities such as supported methods, content types, codecs, and so on.
  • Info : This is used to carry session-related control information such a ISUP, ISDN signaling information.

Now let’s see the 6 availables types of responses:

  • 1XX : Responses in this range are provisional or informational
  • 2XX : Responses indicate a success
  • 3XX : Responses indicate a redirection
  • 4XX : Responses indicate a client error
  • 5XX : Responses indicate a server error
  • 6xx : Responses indicate a global failure

SIP Elements

Here are various elements, definitions of what you can find in SIP world:

  • User Agent (UA) : This an endpoint that can act as both an User Agent Client (UAC) and an User Agent Server (UAS), a good example is a SIP IP Phone.
  • UAC : This is a logical entity that initiates and sends requests, such as those specifiying the INVITE method. the UAC is a logigal role, so it lasts only for the duration of a SIP transaction.
  • UAS : This is an entity that responds to a SIP request by accepting, rejecting, or redirecting the request. The UAS role also lasts  only for the duration of the SIP transaction.
  • Redirect Server : This an (UA) server that provides address translation and redirects clients to alternative destination addresses. It does this by sending 3XX responses to requests.
  • Proxy Server : A SIP proxy server’s primary role is to provide routing, but it can also enforce policies, provide features, and authenticate and authorize users.
  • Registrar Server : Users are registering their current location (location service) so Registrar Server uses the information to provide a lookup service that allows SIP UA to be located.
  • Location Service : This is created by a registrar server and is populated with bindings of Address-of-records (AOR). It is a kind of user’s profiel address. This service can be used by the proxy or redirect servers to retrieve information relating to a called party’s possible locations.

Media Termination Point – MTP

A MTP can bridge tohether two full-duplex voice stram and if necessary convert between G711 ulaw and alaw as well as different sample sizes. So as the MTP bridge is handling each stream independently, H.323 supplementary services can be supported. In other words, MTP will enhance H.323v1 with all these supplementary services(Call Park, Hold , Transfer , Conferencing,…).

Pay attention also that you can use MTP in order to provide the translation between the out-of-band DTMF tones used by SCCP and SIP in-band (payload type) DTMF tones.

SIP CUCM

Since Call Manager 4.X, SIP Trunks can be used for connectivity with SIP Networks and devices via a SIP Proxy Servers.

A SIP Proxy can handle SIP Request , Authentication and Authorization of users and will implement policies and features

MIC,LSC, Security Endpoints

Regarding security endpoints, you have 2 opportunities :

  • Newer phones are using more an existing Mafufacturing Installed Certificate , this is the MIC
  • Meanwhile , old phones will use a Locally Significant Certificate ( LSC) which will be installed by the Certificate Authority Proxy Function (CAPF)

As LSC must be a transaction between the Ip phone and the CAPF , here is the process as it is issued :

  1. IP Phone generates a public/private key pair
  2. A TLS Session is established with the CAPF Service and the keys and identity are sent from the phone to CAPF
  3. The CAPF Service creates and sends an LSC to the phone
  4. The IP Phone installs the LSC

Also for info, the CAPF Service must be in the phone CTL file , which is downloaded from the TFTP when the phone boots .

SIP Transport Layer

SIP is a text-based, application-Layer control signaling protocol that is used for setting up, modifying and tearing down multimedia sessions between participants and takes advantages of elements of HTPP, SMTP and SDP.

SIP Protocol can use  TCP or UDP at the transport layer. The port used is then TCP/UDP/5060 but is TCP/5061 for TLS over TCP

It can use as well multicast as well unicast communication or a combination of the both.

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