ArtDio IPS 1000 user manual Components of SIP

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IPS 1000 Series VoIP Gateway

SIP determines the highest level of common services between the end points. Conferences are established using only the media capabilities that can be supported by all end points.

Determine the availability of the target end point—If a call cannot be completed because the target end point is unavailable, SIP determines whether the called party is already on the phone or did not answer in the allotted number of rings. It then returns a message indicating why the target end point is unavailable.

Establish a session between the originating and target end point—If the call can be completed, SIP establishes a session between the end points. SIP also supports mid-call changes, such as the addition of another end point to the conference or the changing of a media characteristic or Codec.

Handle the transfer and termination of calls—SIP supports the transfer of calls from one end point to another. During a call transfer, SIP simply establishes a session between the transferee and a new end point (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions between all parties.

2.1.1.Components of SIP

SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can function in one of the following roles:

User agent client (UAC)—A client application that initiates the SIP request.

User agent server (UAS)—A server application that contacts the user when a SIP request is received and that returns a response on behalf of the user.

Typically, a SIP end point is capable of functioning as both a UAC and a UAS, but functions only as one or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request.

From an architecture standpoint, the physical components of a SIP network can be grouped into two categories: clients and servers.

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Contents ARTDio Company Inc Version3.1 Update2004/5/7IPS 1000 Series Contents SIP Outbound Authentication IP SettingFAX WEB Management InterfacePreface Safety InstructionsWhat is SIP Components of SIP IPS 1000 Series VoIP Gateway Architecture Panel Descriptions Package ContentsRear Panel IPS 1103 Rear Panel 4 ports IPS 1101 Rear Panel 2 ports LED Indicators IDC Connectors Only for IPS-1000 series 8/16 ports ConnectorsIP Address Information required before InstallationSIP Information Prepare a password for Web ManagementConfirming the Region ID Installation and ConfigurationPhone Setting System console settings Only for 4/8/16 ports IP Address SettingsStatic IP Mode PPPoE Mode Dhcp ModeUse WEB Interface to configure IPS 1000 Series VoIP Gateway Mosa WEBUpon entering the web based configuration interface IPS 1000 Series VoIP Gateway SIP Configuration IPS-1008 Channels and SIP entitySIP Entity SIP Proxy and Register ParametersSIP Outbound Authentication Configure Stun Phone Book Check SIP entity StatusMake SIP Calls Contact Address FXO FXS Other SIP ParametersDialing Plan Call Forward Inbound AuthenticationDevices at two sides are IPS 1000 and the other brands FAXDevices at two sides are all IPS 1000 series gateway Authentication Home Basic General IP Setting AdvancedChannel Phone Book Access Code SIP OutboundBasic / General OFF MACIPS RTPISP IP SettingWEB Dtmf Advanced / GeneralCadence On, off. The on and off SIP Common CMU-PCMA PcmuPcma RFC 2833 Dtmf SIP Outbound Authentication SIP Inbound Anthentication Dialing Plan Dialing Plan Stun Channel IPS 1000 Series VoIP Gateway Phone Book Use Private IP Behind NAT DND Appendix a Phone-Set CommandAppendix IPS 1000 Series VoIP Gateway Enable Appendix B Console CommandVAD, CNG SpecificationsRTP, Rtcp EMIMapping table of characters used in PPPoE USA Region IDTaiwan Contact Information