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Carburettor sip
CONTOH PERMOHONAN SIP DOKTERDeskripsi lengkap
SIP
Marcelo Zanata Components
Error Codes
any endpoint. UA (User Agent) – any UAC (User Agent Client) – UA UA that initialize the call UA that receive the call UAS (User Agent Server) – UA Proxy Server – Do Do call routing, authentication, authentication, authorization, address resolution, loop detection. This can stay int he signaling path or not. UA and Proxy can contact it and get the response with one or more address for the user. Redirect Server – UA Cisco Router can act as it. Registrar Server – Keeps Keeps track of current location location of UA. IOS and CCM can do it. Location Server – maintains maintains the location database of UA B2BUA (Back-to-back User Agent) – a a server acting as UAS and UAC at the same-time, re-initializing the call. CCM can be SIP B2BUA. Presence Server – gather gather presence form Presentities and subscribe information from Watchers
Class of Response Response Code Explanation Informational/ 100 Trying provisional 180 Ringing 181 Call is being forwarded forwarded 182 Queued 183 Session Progress Success 200 OK Redirection Redirection 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service Client-Error 400 Bad Request Methods 401 Unauthorized Cisco gateways can send and receive: 402 Payment Required REGISTER: A UA client sends this message to inform a SIP server of its location. 403 Forbidden INVITE: A caller sends this message to request that another endpoint join a SIP session, such as a 404 Not Found conference conference or a call. This message can also be sent during a call to change session parameters. 405 Method Not Allowed ACK: A SIP UA can receive several several responses to an INVITE. This method acknowledges the final response to 406 Not Acceptable the INVITE. 407 Proxy Auth Required CANCEL: This message ends a call that has not yet been fully established. 408 Request Timeout OPTIONS: This message queries the capabilities of a server. Cisco gateways receive these methods only. 410 Gone BYE: This message ends a session or declines to take a call. 413 Request Entity Too Large Cisco gateway do not generate: 414 Requested URL Too Large INFO: This message is used when data is carried within the message body. 415 Unsupported Media Type PRACK: This message acknowledges receipt of a provisional, or informational, response to a request. 416 Unsupported URI Scheme Scheme REFER This message points to another address to initiate a transfer. 420 Bad Extension SUBSCRIBE This message lets the server know that you want to be notified if a specific event happens. 421 Extension Required NOTIFY This message lets the subscriber know that a specified event has o ccurred. It can also transmit dual 423 Interval Too Brief Brief tone multifrequency (DTMF) tones. 480 Temporarily Not Available UPDATE A UAC uses this to change the session parameters, such as codec used or quality of service (QoS) 481 Transaction Does Does Not Exist settings, before before answering the initial INVITE. 482 Loop Detected SDP fields 483 Too Many Many Hops v: Tells the SDP version 484 Address Incomplete o: Lists the organization of the calling party 485 Ambiguous s: Describes the SDP message 486 Busy Here c: Lists the IP address of the originator 487 Request Terminated : Tells the timer value 488 Not Acceptable Acceptable Here m: Describes the media that the originator expects 491 Request Pending a: Gives the media attributes 493 Undecipherable Server-error 500 Internal Server Error DTMF Relay 501 Not Implemented Named Telephony Events (RFC2833) – RTP Packets with a different type field (In-band) 502 Bad Gateway Key Press Markup Language (KPML) – SIP Subscriber messages with DTMF in XML like format (OOB) 503 Service Unavailable Unsolicited Notify (UN) – SIP Notify messages and without SIP Subscribe (OOB) 504 Server Timeout Cisco RTP – RTP Packets with a different type field. 505 SIP Version Not Supported Call flow with multiple servers 513 Message Too Large Global failure 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere Anywhere 606 Not Acceptable
voice-class sip transport switch udp tcp ” switch from UDP to TCP when a packet gets within 200 bytes of the MTU to avoid UDP fragmentation. “
Other details Default Ports: 5060 TCP/UDP / TLS: 5061 Plain-Text messages Sip address is called URI = uniform resource identifier SIP Dialplan considerations considerations The default behavior of SIP Phone is compare digits to the internal dial plan. When have a match, its sends an INVITE. When you use KPML (Key Press Markup Language), the SIP phone sends each digit to CCM that can instruct the phone what do or route the call.
SIP UA commands sip-ua registrar ipv4:10.30.25.250 tcp registrar ipv4:10.30.25.251 tcp secon sip-server ipv4:10.30.25.252 max-forwards 10 no transport udp
SIP Voice Service commands voice service voip redirect ip2ip sip bind control source-interface lo0 registrar server exp max 1500 min 500
SIP
Marcelo Zanata Early Offer
Delayed Offer
Call flow between two gateways PBX
GWA
GWB
Early Media
Call Flow using a Proxy Server Endpoint Setup
PBX
Setup
SIP Proxy
GW-B
PBX
INVITE
INVITE
Setup
Setup 100 Trying
Call Proceeding 100 Trying
100 Trying
Call Proceeding
Call Proceeding
Alerting
Alerting 180
180 Ringing
Ringing
180 Ringing
Alerting
Connect
Connect 200
200 OK
OK
200 OK
Connect Connect Ack
ACK Connect Ack
ACK Voice
RTP
Connect Ack Voice
RTP
Voice
BYE
Disconnect
Disconnect Release
BYE Release
Disconnect Release
200
OK Release Complete
200 OK
Release Complete
Release Complete
Callmanager acting as B2BUA SIP Phone INVITE, with SDP 100 Trying
CCM
GW-B
INVITE 183 Session Progress, with SDP Session Progress, with SDP 200 OK, with SDP
