4000 Router Series Voice Configuration Guide

SIP messages

SIP is a text-based protocol. A SIP message is either a request from a client to a server, or a response from

a server to a client.

SIP requests include INVITE, ACK, OPTIONS, BYE, CANCEL, and REGISTER.

• INVITE—Invites a user to join a call.

• ACK—Acknowledges the response to a request.

• OPTIONS—Queries for the capabilities.

• BYE—Releases an established call.

• CANCEL—Gives up a call attempt.

• REGISTER—Registers with the SIP registrar.

SIP responses indicate the status of a call or registration. Responses are distinguished by status codes. As

shown in Table 8, eac

h status code is a 3-digit integer, where the first digit defines the class of the

response, and the last two digits describe the response message in more detail.

Table 8 Status codes of responses

Code Descri

tion Class

100–199 The request was received and is being processed. Provisional

200–299 The request was successfully received, understood, and accepted. Success

300–399 A further action needs to be taken to complete the request. Redirection

400–499 The request contains bad syntax or cannot be fulfilled at this server. Client error

500–599 The server failed to fulfill an apparently valid request. Server error

600–699 The request cannot be fulfilled at any server. Global failure

SIP configuration task list

Tasks at a

lance

Configuring SIP UA registration

• (Optional.) Configuring SIP credentials

• (Required.) Enabling a POTS entity to register with the registrar

• (Required.) Configuring registrar information

(Required.) Use one of the following methods to specify a call destination address for a VoIP entity

• Configuring the call destination IP address for a VoIP entity

• Configuring a VoIP entity to obtain the call destination address from a proxy server

• Configuring the destination domain name and port number for a VoIP entity

(Optional.) Configuring extended SIP functions

• Configuring out-of-band DTMF

• Configuring periodic refresh of SIP sessions

• Configuring PSTN cause-to-SIP status mappings

• Configuring caller privacy

• Setting the P-Asserted-Identity or P-Preferred-Identity header field