SIP IP電話系統定義了註冊/登出、呼叫、重定向、能力查詢四種工作過程。

註冊/登出過程

　　SIP為使用者定義了註冊和登出過程，其目的是可以動態建立使用者的邏輯地址和其當前聯絡地址之間的對應關係，以方便實現呼叫路由和對使用者移動性的 支援。邏輯地址和聯絡地址的分離也方便了使用者，它不論在何處、使用何種裝置，都可以通過唯一的邏輯地址進行通訊。

　　註冊/登出過程是通過REGISTER訊息和200成功響應來實現的。在註冊/登出時，使用者將其邏輯地址和當前聯絡地址通過REGISTER消 息傳送給其註冊伺服器，註冊伺服器對該請求訊息進行處理，並以200成功響應訊息通知使用者註冊/登出成功。

例 

REGISTER sip:192.168.0.157 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.212;rport;branch=z9hG4bKc0a800d4000000104c21966c00007c9900000004
Content-Length: 0
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 REGISTER
From: <sip:[email protected]>;tag=1525750021180
Max-Forwards: 70
To: <sip:[email protected]>
User-Agent: SJphone/1.60.289a (SJ Labs)
Authorization: Digest username="8002",realm="asterisk",nonce="4fc62d23",uri="sip:192.168.0.157",response="39677e3edaed7037bd8af592d8d2a038",algorithm="MD5"

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.212;branch=z9hG4bKc0a800d4000000104c21966c00007c9900000004;received=192.168.0.212;rport=5060
From: <sip:[email protected]>;tag=1525750021180
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:

[email protected]>
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.212;branch=z9hG4bKc0a800d4000000104c21966c00007c9900000004;received=192.168.0.212;rport=5060
From: <sip:[email protected]>;tag=1525750021180
To: <sip:[email protected]>;tag=as3d657ebf
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:[email protected]:5060>;expires=120
Date: Wed, 23 Jun 2010 05:07:00 GMT
Content-Length: 0

呼叫過程

　　SIP IP電話系統中的呼叫是通過INVITE邀請請求、成功響應和ACK確認請求的三次握手來實現的。即當主叫使用者代理要發起呼叫時，它構造一個INVITE 訊息，併發送給被叫。被叫收到邀請後決定接受該呼叫，就回送一個成功響應（狀態碼為200）。主叫方收到成功響應後，向對方傳送ACK請求。被叫收到 ACK請求後，呼叫成功建立。

　　呼叫的終止通過BYE請求訊息來實現。當參與呼叫的任一方要終止呼叫時，它就構造一個BYE請求訊息，併發送給對方。對方收到BYE請求後，釋 放與此呼叫相關的資源，回送一個成功響應，表示呼叫已經終止。

　　當主被叫雙方已建立呼叫，如果任一方想要修改當前的通訊引數（通訊型別、編碼等），可以通過傳送一個對話內的INVITE請求訊息（稱為re- INVITE）來實現。

例

INVITE sip:[email protected]:49152 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK0083a186;rport
From: "8002" <sip:[email protected]>;tag=as044943cc
To: <sip:[email protected]:49152>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 23 Jun 2010 05:13:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 363

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.212;branch=z9hG4bKc0a800d40000002b4c2197e00000549c0000004a;received=192.168.0.212;rport=5060
From: "unknown"<sip:[email protected]>;tag=156294689213
To: <sip:[email protected]>;tag=as46e231f6
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK0083a186;rport=5060;received=192.168.0.157
From: "8002" <sip:[email protected]>;tag=as044943cc
To: "unknown" <sip:[email protected]:49152>;tag=653819a85587
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)

OPTIONS sip:192.168.0.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.212;rport;branch=z9hG4bKc0a800d4000000104c2197e100000a7a0000004d
Content-Length: 0
Call-ID: [email protected]
CSeq: 18 OPTIONS
From: <sip:[email protected]>;tag=1563048416840
Max-Forwards: 70
To: <sip:192.168.0.157>

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.212;branch=z9hG4bKc0a800d4000000104c2197e100000a7a0000004d;received=192.168.0.212;rport=5060
From: <sip:[email protected]>;tag=1563048416840
To: <sip:192.168.0.157>;tag=as3aafc1a7
Call-ID: [email protected]
CSeq: 18 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:192.168.0.157>
Accept: application/sdp
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK0083a186;rport=5060;received=192.168.0.157
From: "8002" <sip:[email protected]>;tag=as044943cc
To: "unknown" <sip:[email protected]:49152>;tag=653819a85587
Contact: <sip:[email protected]:49152>
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK0083a186;rport=5060;received=192.168.0.157
From: "8002" <sip:[email protected]>;tag=as044943cc
To: "unknown" <sip:[email protected]:49152>;tag=653819a85587
Contact: <sip:[email protected]:49152>
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 271
Content-Type: application/sdp
Server: SJphone/1.65.377a (SJ Labs)
Supported: replaces,norefersub,timer

ACK sip:[email protected]:49152 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.157:5060;branch=z9hG4bK4ed93e8f;rport
From: "8002" <sip:[email protected]>;tag=as044943cc
To: <sip:[email protected]:49152>;tag=653819a85587
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

重定向過程

　　當重定向伺服器（其功能可包含在代理伺服器 和使用者終端中）收到主叫使用者代理的INVITE邀請訊息，它通過查詢定位伺服器發現該呼叫應該被重 新定向（重定向的原因有多種，如使用者位置改變、實現負荷分擔等等），就構造一個重定向響應訊息（狀態碼為3xx），將新的目標地址回送給主叫使用者代理。主 叫使用者代理收到重定向響應訊息後，將逐一向新的目標地址傳送INVITE邀請，直至收到成功響應並建立呼叫。如果嘗試了所有的新目標而無法建立呼叫，則本 次呼叫失敗。

能力查詢過程

　　SIP IP電話系統還提供了一種讓使用者在不打擾對方使用者的情況下查詢對方通訊能力的手段。可查詢的內容包括：對方支援的請求方法（methods）、支援的內容 型別、支援的擴充套件項、支援的編碼等等。

　　能力查詢通過OPTION請求訊息來實現。當用戶代理想要查詢對方的能力時，它構造一個OPTION請求訊息，傳送給對方。對方收到該請求訊息 後，將自己支援的能力通過響應訊息回送給查詢者。如果此時自己可以接收呼叫，就傳送成功響應（狀態碼為200），如果此時自己忙，就傳送自身忙響應（狀態 碼為 486）。因此，能力查詢過程也可以用於查詢對方的忙閒狀態，看是否能夠接受呼叫。