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SIP INVITEメソッドで200レスポンスを継続的に取得

2016-04-28 9 views
-1

INVITEリクエストの被呼者が私に200（OK）レスポンスを送信していますが、ACKを送信していますが、今は着信が肯定応答を受信して​​いないかどうかわかりません私は依然として200の応答を継続して受けています。以下は要求と応答です。ここでSIP INVITEメソッドで200レスポンスを継続的に取得

は全体のSIPダイアログです：

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: [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]92.168.1.101:5060", response="b1e38ea10061a0224e2189e9177fff1c", algorithm=MD5 
User-Agent: MySIP V2.0 
Content-Length: 0

出典

2016-04-28 Sahitya kanral

+0

あなたはそれをキャプチャして、なぜヘッダ経由で見ることができますか？ – piyushj

答えて

0

私はあなたの第二に、あなたが同じCall-ID、タグからの再使用していて、オリジナルのINVITEとしてタグにINVITEリクエストをするので、問題があると思います。 2番目のINVITE要求は、新しいトランザクションであるため、すべて異なる値にする必要があります。第二に

出典

2016-04-28 22:45:05 sipwiz

0

は、INVITE：

  • タグから異なるはず
  • 経由の分岐が異なっている必要がありますのためのACKにも

を逃すべきであるタグに

  • 200 OK：

    • From、 To、Via、Call-IDは200 OKと同じにする必要があります。

    それ以外の場合は、着信側でも明らかなバグがあります（INVITE with toタグを受け入れるなど）。

    • 出典

    2016-04-29 11:12:13 Istvan

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