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在SIP INVITE方法中獲得200響應連續

2016-04-28 686 views
-1

在一個INVITE請求中，被叫發送一個200（OK）響應，我發送一個ACK，但現在我不知道被叫是否沒有收到確認，我仍然在不斷收到200個迴應，下面是請求和迴應。在SIP INVITE方法中獲得200響應連續

這裏是整個SIP對話：

INVITE sip:[email protected] SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bKe446822249352251a5bd13b6e66ef303;rport 
Max-Forwards: 70 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 1 INVITE 
Contact: <sip:[email protected]:5070> 
Allow: INVITE,ACK,BYE,CANCEL 
User-Agent: MySIP V2.0 
Content-Type: application/sdp 
Content-Length: 179 
<-------------> 


<--- Reliably Transmitting (no NAT) to 192.168.1.9:5070 ---> 
SIP/2.0 401 Unauthorized 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bKe446822249352251a5bd13b6e66ef303;received=192.168.1.9;rport=5070 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 1 INVITE 
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH 
Supported: replaces, timer 
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50310a1a" 
Content-Length: 0 



<--- SIP read from UDP:192.168.1.9:5070 ---> 
ACK sip:[email protected] SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bKe446822249352251a5bd13b6e66ef303;rport 
Max-Forwards: 70 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 1 ACK 
User-Agent: MySIP V2.0 
Content-Length: 0 

<-------------> 


<--- SIP read from UDP:192.168.1.9:5070 ---> 
INVITE sip:[email protected] SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bK4fc5c00de7440b65f8b5355cfacb82d2;rport 
Max-Forwards: 70 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 2 INVITE 
Contact: <sip:[email protected]:5070> 
Allow: INVITE,ACK,BYE,CANCEL 
User-Agent: MySIP V2.0 
Content-Type: application/sdp 
Content-Length: 179 
Authorization: Digest username="3001", realm="asterisk", nonce="50310a1a", opaque="", uri="sip:[email protected]:5060", response="b1e38ea10061a0224e2189e9177fff1c", algorithm=MD5 
<-------------> 


<--- Transmitting (no NAT) to 192.168.1.9:5070 ---> 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bK4fc5c00de7440b65f8b5355cfacb82d2;received=192.168.1.9;rport=5070 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: b6c72c48-402d-40d3-aabf-0538[email protected] 
CSeq: 2 INVITE 
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH 
Supported: replaces, timer 
Contact: <sip:[email protected]:5060> 
Content-Length: 0 
<------------> 


<--- Transmitting (no NAT) to 192.168.1.9:5070 ---> 
SIP/2.0 180 Ringing 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bK4fc5c00de7440b65f8b5355cfacb82d2;received=192.168.1.9;rport=5070 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 2 INVITE 
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH 
Supported: replaces, timer 
Contact: <sip:[email protected]:5060> 
Content-Length: 0 
<------------> 

<--- Reliably Transmitting (no NAT) to 192.168.1.9:5070 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bK4fc5c00de7440b65f8b5355cfacb82d2;received=192.168.1.9;rport=5070 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 2 INVITE 
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH 
Supported: replaces, timer 
Contact: <sip:[email protected]:5060> 
Content-Type: application/sdp 
Content-Length: 195 
<------------> 


<--- SIP read from UDP:192.168.1.9:5070 ---> 
ACK sip:[email protected]:5060 SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bKd8c26ff688480593ce616e913ac8e609;rport 
Max-Forwards: 70 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 2 ACK 
Authorization: Digest username="3001", realm="asterisk", nonce="50310a1a", opaque="", uri="sip:[email protected]:5060", response="b1e38ea10061a0224e2189e9177fff1c", algorithm=MD5 
User-Agent: MySIP V2.0 
Content-Length: 0 
<-------------> 


Retransmitting #1 (no NAT) to 192.168.1.9:5070: 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bK4fc5c00de7440b65f8b5355cfacb82d2;received=192.168.1.9;rport=5070 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 2 INVITE 
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH 
Supported: replaces, timer 
Contact: <sip:[email protected]:5060> 
Content-Type: application/sdp 
Content-Length: 195 
--- 

<--- SIP read from UDP:192.168.1.9:5070 ---> 
ACK sip:[email protected]:5060 SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.9:5070;branch=z9hG4bK542e162569381a2bd29c111fa3ea6e03;rport 
Max-Forwards: 70 
From: Sahitya<sip:[email protected]>;tag=m1RbnnhW7m 
To: Prithvi<sip:[email protected]>;tag=as69cbf848 
Call-ID: [email protected] 
CSeq: 2 ACK 
Authorization: Digest username="3001", realm="asterisk", nonce="50310a1a", opaque="", uri="sip:[email protected]:5060", response="b1e38ea10061a0224e2189e9177fff1c", algorithm=MD5 
User-Agent: MySIP V2.0 
Content-Length: 0

來源

2016-04-28 Sahitya kanral

+0

你在哪裏捕捉它，爲什麼我可以通過頭部看到？ – piyushj

回答

0

我懷疑問題是因爲你的第二個INVITE請求你是相同的Call-ID重新使用，從標籤和標記的原始邀請。對於第二個INVITE請求，它們應該都是不同的值，因爲它是一個新的事務。

來源

2016-04-28 22:45:05 sipwiz

0

在第二邀請：

  • 的從標籤應該是不同的
  • 威盛分支應該是不同的
  • 收件人標記應該丟失

此外，在ACK的200 OK：

  • the From， To，Via和Call-ID應與200相同

否則在被叫方也有明顯的錯誤（例如接受帶有標記的INVITE）。

來源

2016-04-29 11:12:13 Istvan

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