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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 03 2008 at 10:18am | IP Logged
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Hi,
It seems Lanscape does not send any ACK when it receives SIP "487 Request Terminated" message from Asterisk. This message is sent in following scenario.
Code:
1. (A)--------------------INVITE --------------------->(B)
2. (A)<------------------180 RIGING------------------(B)
3. (A)--------------------CANCEL------------------->(B)
4. (A)<------------------OK----------------------------(B)
5. (A)<---------487 Request Terminated-----------(B)
6. (A)--------------------ACK------------------------->(B)
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OK (4) reply corresponds to the CANCEL (3) request.
487 Request Terminated (5) reply corresponds to the INVITE (1) request.
If LMEVoip do not send the ACK asterisk tries to resend 487 message every 2 seconds for 4 times. Asterisk will not respond to any other invites during these 8 seconds.
Regards,
Jalal Abedinejad
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 05 2008 at 1:36pm | IP Logged
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Hi Jalal,
We hope you are well. We will look into it right now. The last time we tested against Asterisk, all was OK.
Question 1:
What version and distribution of Asterisk are you using?
It must be a simple SIP interop problem. We know the media engine is designed to send the final ACK. Here is a log from a media engine soft phone talking to our Centrex SIP proxy that generates the call scenario you posted:
Code:
************* Log Opened (May 05 13:29:48) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#3, 46562 Ms, To: 192.168.1.2:5060) >>>>
INVITE sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK00e5e32c
From: "Test Phone" <sip:333@ps>;tag=e5d6d3;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Contact: <sip:333@192.168.1.2:5068>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086332 INVITE
Max-Forwards: 70
Organization: 2456E5FD-DC94-417D-BDF7-55A654EFB9E5
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 244
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=333 15062468 15062468 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 20000 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#3, 46578 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5068;rport=5068;branch=z9hG4bK00e5e32c;received=192.168.1.2
From: "Test Phone" <sip:333@ps>;tag=e5d6d3;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Call-ID: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086332 INVITE
Server: LanScape Centrex Proxy/3.42.2.8 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#4, 0 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5068;rport=5068;branch=z9hG4bK00e5e32c;received=192.168.1.2
From: "Test Phone" <sip:333@ps>;tag=e5d6d3;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Call-ID: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086332 INVITE
Proxy-Authenticate: Digest realm="ps", nonce="ee9f63726ff67019386fcdb98b658802",
opaque="28b445a768e6194531df00bba2d8dd68"
Server: LanScape Centrex Proxy/3.42.2.8 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#4, 32 Ms, To: 192.168.1.2:5060) >>>>
ACK sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;received=192.168.1.2;rport=5068;branch=z9hG4bK00e5e32c
From: "Test Phone" <sip:333@ps>;tag=e5d6d3;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Call-Id: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086332 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#5, 15 Ms, To: 192.168.1.2:5060) >>>>
INVITE sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK00e5e34d
From: "Test Phone" <sip:333@ps>;tag=e5d6f4;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Contact: <sip:333@192.168.1.2:5068>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086379 INVITE
Max-Forwards: 70
Organization: 2456E5FD-DC94-417D-BDF7-55A654EFB9E5
Proxy-Authorization: Digest algorithm=md5,nonce="ee9f63726ff67019386fcdb98b658802",
opaque="28b445a768e6194531df00bba2d8dd68",realm="ps",response="179de05dee676b554c42164d91880456",
uri="sip:111@ps",username="guest"
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 244
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=333 15062500 15062500 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 20000 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#5, 31 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5068;rport=5068;branch=z9hG4bK00e5e34d;received=192.168.1.2
From: "Test Phone" <sip:333@ps>;tag=e5d6f4;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Call-ID: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086379 INVITE
Server: LanScape Centrex Proxy/3.42.2.8 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#6, 172 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5068;received=192.168.1.2;rport=5068;branch=z9hG4bK00e5e34d
From: "Test Phone" <sip:333@ps>;tag=e5d6f4;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>;tag=34891c40
Call-ID: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086379 INVITE
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.8 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#6, 2000 Ms, To: 192.168.1.2:5060) >>>>
CANCEL sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK00e5e34d
From: "Test Phone" <sip:333@ps>;tag=e5d6f4;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>;tag=34891c40
Call-Id: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086379 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#7, 1828 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5068;rport=5068;branch=z9hG4bK00e5e34d;received=192.168.1.2
From: "Test Phone" <sip:333@ps>;tag=e5d6f4;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>;tag=34891c40
Call-ID: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086379 CANCEL
Server: LanScape Centrex Proxy/3.42.2.8 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#8, 16 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 487 Transaction Cancelled
Via: SIP/2.0/UDP 192.168.1.2:5068;received=192.168.1.2;rport=5068;branch=z9hG4bK00e5e34d
From: "Test Phone" <sip:333@ps>;tag=e5d6f4;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Call-ID: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086379 INVITE
Proxy-Authenticate: Digest realm="ps", nonce="ee9f63726ff67019386fcdb98b658802",
opaque="5510f30ab5e3ba1f598971350ea81768"
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.8 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#7, 16 Ms, To: 192.168.1.2:5060) >>>>
ACK sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;received=192.168.1.2;rport=5068;branch=z9hG4bK00e5e34d
From: "Test Phone" <sip:333@ps>;tag=e5d6f4;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:111@ps>
Call-Id: 214c51eb-275f-479e-ae9f-39761c3fc1c6-00000c40@192.168.1.2
CSeq: 15086379 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
************* Log Closed (May 05 13:30:49) *************
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 05 2008 at 2:07pm | IP Logged
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Jalal,
Here is the same test against Asterisk v1.4.9 and it looks OK. We will test again when we receive the Asterisk version and distribution you are using.
Code:
************* Log Opened (May 05 14:02:45) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#14, 15078 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:201@lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5094;rport;branch=z9hG4bK01039540
From: "Extension 200" <sip:200@lslab.com>;tag=103c8f3;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>
Contact: <sip:200@192.168.1.2:5094>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229169 INVITE
Max-Forwards: 70
Organization: 2456E5FD-DC94-417D-BDF7-55A654EFB9E5
x-CustomHeader-Extension-200: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 220
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=200 17002781 17002781 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8666 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#20, 15078 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5094;branch=z9hG4bK01039540;received=192.168.1.2;rport=5094
From: "Extension 200" <sip:200@lslab.com>;tag=103c8f3;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>;tag=as202efeef
Call-ID: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229169 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ee9d24f"
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#15, 32 Ms, To: 192.168.1.122:5060) >>>>
ACK sip:201@lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5094;received=192.168.1.2;rport=5094;branch=z9hG4bK01039540
From: "Extension 200" <sip:200@lslab.com>;tag=103c8f3;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>;tag=as202efeef
Call-Id: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229169 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-200: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#16, 0 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:201@lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5094;rport;branch=z9hG4bK01039570
From: "Extension 200" <sip:200@lslab.com>;tag=103c923;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>
Contact: <sip:200@192.168.1.2:5094>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229216 INVITE
Max-Forwards: 70
Organization: 2456E5FD-DC94-417D-BDF7-55A654EFB9E5
Proxy-Authorization: Digest algorithm=md5,nonce="1ee9d24f",realm="asterisk",
response="6ad4e2c74881a3de43f6697eccd5e4f4",uri="sip:201@lslab.com",username="200"
x-CustomHeader-Extension-200: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 220
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=200 17002828 17002828 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8666 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#21, 47 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5094;branch=z9hG4bK01039570;received=192.168.1.2;rport=5094
From: "Extension 200" <sip:200@lslab.com>;tag=103c923;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>
Call-ID: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201@192.168.1.122>
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#22, 94 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5094;branch=z9hG4bK01039570;received=192.168.1.2;rport=5094
From: "Extension 200" <sip:200@lslab.com>;tag=103c923;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>;tag=as67832485
Call-ID: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201@192.168.1.122>
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#17, 2187 Ms, To: 192.168.1.122:5060) >>>>
CANCEL sip:201@lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5094;rport;branch=z9hG4bK01039570
From: "Extension 200" <sip:200@lslab.com>;tag=103c923;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>;tag=as67832485
Call-Id: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229216 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-200: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#23, 2062 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.2:5094;branch=z9hG4bK01039570;received=192.168.1.2;rport=5094
From: "Extension 200" <sip:200@lslab.com>;tag=103c923;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>;tag=as67832485
Call-ID: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#18, 16 Ms, To: 192.168.1.122:5060) >>>>
ACK sip:201@lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5094;received=192.168.1.2;rport=5094;branch=z9hG4bK01039570
From: "Extension 200" <sip:200@lslab.com>;tag=103c923;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>;tag=as67832485
Call-Id: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229216 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-200: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#24, 31 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5094;branch=z9hG4bK01039570;received=192.168.1.2;rport=5094
From: "Extension 200" <sip:200@lslab.com>;tag=103c923;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:201@lslab.com>;tag=as67832485
Call-ID: a1ccd0c4-c829-4678-8754-19a61e168e43-000016d4@192.168.1.2
CSeq: 229216 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201@192.168.1.122>
Content-Length: 0
************* Log Closed (May 05 14:02:59) *************
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 06 2008 at 1:22am | IP Logged
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Hi,
We are using Asterisk v1.4.17-1 . You were right about LMEVoip sending the Ack in response to 487 Request Terminated. But LMEVoip sometimes sends the ACK and sometimes does not send it. I checked the situation when LMEVoip does not send the ACK and I found following issue. Here are two different calls that LMEVoip did not send the ACK.
Code:
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#28, 9984 Ms, To: 10.10.10.155:5060) >>>>
INVITE sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;rport;branch=z9hG4bK1d6628a9
From: 1001 <sip:300@10.10.10.155>;tag=1d66728d
To: <sip:260@10.10.10.155>
Contact: <sip:300@10.10.10.157:5060>
Call-Id: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 INVITE
Max-Forwards: 70
Organization: 2E6A60B4-035E-4926-BA71-733FBF180176
Content-Length: 224
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 493221828 493221828 IN IP4 10.10.10.157
s=LanScape
c=IN IP4 10.10.10.157
t=0 0
m=audio 8002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#28, 9984 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d6628a9;received=10.10.10.157;rport=5060
From: 1001 <sip:300@10.10.10.155>;tag=1d66728d
To: <sip:260@10.10.10.155>;tag=as5f885eb9
Call-ID: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67d4991f"
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#29, 0 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;received=10.10.10.157;rport=5060;branch=z9hG4bK1d6628a9
From: 1001 <sip:300@10.10.10.155>;tag=1d66728d
To: <sip:260@10.10.10.155>;tag=as5f885eb9
Call-Id: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#30, 0 Ms, To: 10.10.10.155:5060) >>>>
INVITE sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;rport;branch=z9hG4bK1d6628b9
From: 1001 <sip:300@10.10.10.155>;tag=1d66729d
To: <sip:260@10.10.10.155>
Contact: <sip:300@10.10.10.157:5060>
Call-Id: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 INVITE
Max-Forwards: 70
Organization: 2E6A60B4-035E-4926-BA71-733FBF180176
Proxy-Authorization: Digest algorithm=md5,nonce="67d4991f",realm="asterisk",
response="7cc6a1987d59ab0f4ff31d83b56de928",uri="sip:260@10.10.10.155",username="300"
Content-Length: 224
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 493221843 493221843 IN IP4 10.10.10.157
s=LanScape
c=IN IP4 10.10.10.157
t=0 0
m=audio 8002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#31, 20016 Ms, To: 10.10.10.155:5060) >>>>
CANCEL sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;rport;branch=z9hG4bK1d6628b9
From: 1001 <sip:300@10.10.10.155>;tag=1d66729d
To: <sip:260@10.10.10.155>
Call-Id: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#29, 20016 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d6628a9;received=10.10.10.157;rport=5060
From: 1001 <sip:300@10.10.10.155>;tag=1d66728d
To: <sip:260@10.10.10.155>;tag=as5f885eb9
Call-ID: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#30, 16 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d6628b9;received=10.10.10.157;rport=5060
From: 1001 <sip:300@10.10.10.155>;tag=1d66729d
To: <sip:260@10.10.10.155>;tag=as5f885eb9
Call-ID: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:260@10.10.10.155>
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#31, 1265 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d6628a9;received=10.10.10.157;rport=5060
From: 1001 <sip:300@10.10.10.155>;tag=1d66728d
To: <sip:260@10.10.10.155>;tag=as5f885eb9
Call-ID: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#32, 1453 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d6628a9;received=10.10.10.157;rport=5060
From: 1001 <sip:300@10.10.10.155>;tag=1d66728d
To: <sip:260@10.10.10.155>;tag=as5f885eb9
Call-ID: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#33, 2719 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d6628a9;received=10.10.10.157;rport=5060
From: 1001 <sip:300@10.10.10.155>;tag=1d66728d
To: <sip:260@10.10.10.155>;tag=as5f885eb9
Call-ID: 223052f7-92d2-42c6-9bc5-f51a282bef95-00000db4@10.10.10.157
CSeq: 6696077 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
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Code:
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#29, 9797 Ms, To: 10.10.10.155:5060) >>>>
INVITE sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;rport;branch=z9hG4bK1d75e685
From: 229 <sip:300@10.10.10.155>;tag=1d75c3eb
To: <sip:260@10.10.10.155>
Contact: <sip:300@10.10.10.157:5060>
Call-Id: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 INVITE
Max-Forwards: 70
Organization: 2E6A60B4-035E-4926-BA71-733FBF180176
Content-Length: 224
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 494239078 494239078 IN IP4 10.10.10.157
s=LanScape
c=IN IP4 10.10.10.157
t=0 0
m=audio 8010 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#30, 9782 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d75e685;received=10.10.10.157;rport=5060
From: 229 <sip:300@10.10.10.155>;tag=1d75c3eb
To: <sip:260@10.10.10.155>;tag=as20695f17
Call-ID: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04c81247"
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#30, 0 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;received=10.10.10.157;rport=5060;branch=z9hG4bK1d75e685
From: 229 <sip:300@10.10.10.155>;tag=1d75c3eb
To: <sip:260@10.10.10.155>;tag=as20695f17
Call-Id: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#31, 15 Ms, To: 10.10.10.155:5060) >>>>
INVITE sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;rport;branch=z9hG4bK1d75e686
From: 229 <sip:300@10.10.10.155>;tag=1d75c3ec
To: <sip:260@10.10.10.155>
Contact: <sip:300@10.10.10.157:5060>
Call-Id: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 INVITE
Max-Forwards: 70
Organization: 2E6A60B4-035E-4926-BA71-733FBF180176
Proxy-Authorization: Digest algorithm=md5,nonce="04c81247",realm="asterisk",
response="18ef68a4a6838209ee3155f04938680c",uri="sip:260@10.10.10.155",username="300"
Content-Length: 224
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 494239078 494239078 IN IP4 10.10.10.157
s=LanScape
c=IN IP4 10.10.10.157
t=0 0
m=audio 8010 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#32, 3000 Ms, To: 10.10.10.155:5060) >>>>
CANCEL sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5060;rport;branch=z9hG4bK1d75e686
From: 229 <sip:300@10.10.10.155>;tag=1d75c3ec
To: <sip:260@10.10.10.155>
Call-Id: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#31, 3015 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d75e685;received=10.10.10.157;rport=5060
From: 229 <sip:300@10.10.10.155>;tag=1d75c3eb
To: <sip:260@10.10.10.155>;tag=as20695f17
Call-ID: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#32, 16 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d75e686;received=10.10.10.157;rport=5060
From: 229 <sip:300@10.10.10.155>;tag=1d75c3ec
To: <sip:260@10.10.10.155>;tag=as20695f17
Call-ID: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:260@10.10.10.155>
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#33, 1266 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d75e685;received=10.10.10.157;rport=5060
From: 229 <sip:300@10.10.10.155>;tag=1d75c3eb
To: <sip:260@10.10.10.155>;tag=as20695f17
Call-ID: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#34, 1265 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d75e685;received=10.10.10.157;rport=5060
From: 229 <sip:300@10.10.10.155>;tag=1d75c3eb
To: <sip:260@10.10.10.155>;tag=as20695f17
Call-ID: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#35, 2563 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5060;branch=z9hG4bK1d75e685;received=10.10.10.157;rport=5060
From: 229 <sip:300@10.10.10.155>;tag=1d75c3eb
To: <sip:260@10.10.10.155>;tag=as20695f17
Call-ID: c9b5416a-b7e6-4c69-96b2-041996cadbbe-00001140@10.10.10.157
CSeq: 7708006 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
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Sometimes asterisk does not send "SIP Trying" or any other message in response to LMEVoip Invite (which includes authentication params) . This maybe is because of not receiving this packet at all by asterisk or maybe because of invalid authentication params. Anyhow LMEVoip does not try to resend the invite message (I don't know what RFC says about resending invite message in such situation!) and after a short time we cancel the call. After Cancel is sent in such situation if Asterisk responses with "487 Request Terminated" message LMEVoip does not send the ACK.
Thanks,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 06 2008 at 6:31am | IP Logged
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Got it. Good clarification. we will continue to investigate...
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 06 2008 at 11:39am | IP Logged
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Jalal,
Can you give us two extension access to your Asterisk box? We are testing latest media engine against Asterisk v1.4.19.1 and cannot reproduce. It would be easier to access your Asterisk v1.4.17-1 instead of us blowing alot of time configuring another box.
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 06 2008 at 11:46am | IP Logged
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.. or maybe point us in the directin of a Asterisk v1.4.17 LiveCd image or a VmWare workstation v5 image.
Just a thought....
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 06 2008 at 2:32pm | IP Logged
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Hi Jalal,
Forget the last two posts. We are pretty sure we have identified the cause. The media engine internally waits for a fixed amount of time to receive CANCEL associated responses. If this hard coded time is reached (2000 Ms), the call will be terminated but the responses (even if received after the timeout) will be ignored. That is why the media engine is not sending out the final INVITE ACK. Event thought the SIP logs show that CANCEL responses are received, the cancel timeout is most likely being violated.
We added a new API procedure SetCallCancelTimeout() so apps can adjust CANCEL processing time outs. Here is an excerpt from upcoming release notes:
------------------------------------------------------------ -------
The SetCallCancelTimeout() API procedure has been added to allow applications to control how long the media engine will wait for outgoing call CANCEL responses from the far end of the call. Prior to this release, the media engine used a hard coded timeout value.
When the media engine transmits a CANCEL request for an outgoing call, it expects to receive a "200 OK" response for the transmitted CANCEL request and a "487 Transaction Terminated" response for the initial INVITE request. Note that the "487 Transaction Terminated" response received will also be answered by a final ACK from the media engine.
The media engine will wait the "CallCancelTimeout" as set by the API procedure for the "200 OK" response. It will also wait the "CallCancelTimeout" for the "487 Transaction Terminated" response. The order of these responses does not matter. If the media engine does not receive both expected responses, the phone call will proceed with cancel termination. If the responses arrive after the timeout as set by the API procedure, the responses will be ignored.
------------------------------------------------------------ -------
This update will be made available in the v6 product release.
Support
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 06 2008 at 11:31pm | IP Logged
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Hi,
I think your thought of CancelTimeout problem is not true. I checked both Call Logs I have sent in previous post and realized that there is no Cancel Timeout problem.
Code:
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#28, 9984 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#29, 0 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@10.10.10.155 SIP/2.0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#30, 0 Ms, To: 10.10.10.155:5060) >>>>
INVITE sip:260@10.10.10.155 SIP/2.0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#31, 20016 Ms, To: 10.10.10.155:5060) >>>>
CANCEL sip:260@10.10.10.155 SIP/2.0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#29, 20016 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#30, 16 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 200 OK
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Something bad about your logs is that you don't log exact time of receiving/sending the messages. But at least you have a parameter in each log that says amount of time passed from last message in the same direction. So if you check above call messages, if we suppose Rx#28, Tx#29 and Tx#30 are received/sent in exact the same time (This supposition is by sure correct), message Tx#31 (CANCEL) is sent 20016 (ms) later and also Rx#29(Request Terminated) is received 20016 (ms) passed from Rx#28 which is in exact the same time as Tx#29 , Tx#30. So Rx#29 (Request Terminated) is received 0 (ms) passed of Tx#31 (CANCEL). Again Rx#30 (OK) is received 16 (ms) passed of Rx#29 so is received 16 (ms) passed of Tx#31 (Cancel).
So you see your supposition of Cacnel 2000 (ms) timeout problem is completely wrong. If you recheck the second call log you will see the same results.
Adding SetCallCancelTimeout is good but will not solve this problem.
Another problem that you have not noticed is that why asterisk have not sent us SIP Trying in response to our invite message. I'm pretty sure that it has received this invite message because they are on the same LAN. I think there is something in the second invite message that causes asterisk not to reply.
If you want I can prepare you a VPN connection to our site for your test. Contact me directly about this.
Regards,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 07 2008 at 6:05am | IP Logged
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Jalal,
As far as the timing in the SIP logs, the “time stamps” that are logged are the relative UDP transmit and receive times between Rx and Tx SIP messages. The times that appear in the logs are not associated with when the media engine gets around to processing the received SIP messages. They are simple transmit and receive relative times.
Because we have no way of knowing what end users are doing with the media engine and how much system loading the media engine is experiencing, we have to make an educated decision based on whatever information has been brought to our attention.
So to say we are wrong is inaccurate. The only way the media engine would not sent a final ACK to a CANCELed call after receiving a “487 Transaction Terminated” is if it times out waiting for the “487 Transaction Terminated” for the original INVITE request. We are not guessing when we say this. That appears to be the only thing that could be occurring based on the source code for the media engine. Are there other possibilities? Maybe. We have overlooked things in the past and have been wrong before.
We have tested against latest v1.4.19.1 of asterisk and do not see this issue. Sounds like a bug in asterisk. If you have free time, break out your debugger and see why v1.4.17-1 of Asterisk is not ‘sometimes” sending provisional responses. If we work the problem from both ends, maybe we can find an additional issue. Let us know what you find.
Many thanks for your offer for the VPN connection. We will make a note of it and take you up on your offer when we have more time.
Support
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 07 2008 at 10:04am | IP Logged
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Hi,
I added some logs to our log system and checked the timing you said. And still I see the same results as I said in my previous post. Something I have forgot to tell you was that TerminateCall API function which sends the SIP Cancel message has a Timeout parameter which I have set it to 1000 ms. And as you can see in following log in such situation which LMEVoip does not send Ack in reply to 487 message, this function returns SIPCallTimeout error.
Code:
11:58:38.578[f74]SIP::Phone Line Notification:SipOutgoingCallStart
11:58:38.593[f74]SIP::Phone Line Notification:SipDialTone
11:58:38.593[f74]SIP::Phone Line Notification:SipDialing
11:58:38.593[f74]SIP::Phone Line Notification:SipSendInvite
11:58:38.593[f74]SIP::Tx:INVITE sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5061;rport;branch=z9hG4bK00b777c9
From: 229 <sip:300@10.10.10.155>;tag=b7d4f3
To: <sip:260@10.10.10.155>
Contact: <sip:300@10.10.10.157:5061>
Call-Id: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 INVITE
Max-Forwards: 70
Organization: 2E6A60B4-035E-4926-BA71-733FBF180176
Content-Length: 222
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 12018140 12018140 IN IP4 10.10.10.157
s=LanScape
c=IN IP4 10.10.10.157
t=0 0
m=audio 8006 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
11:58:38.593[f74]SIP::Phone Line Notification:SipStartOutgoingRing
11:58:38.593[eb8]SIP::Rx:SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.157:5061;branch=z9hG4bK00b777c9;received=10.10.10.157;rport=5061
From: 229 <sip:300@10.10.10.155>;tag=b7d4f3
To: <sip:260@10.10.10.155>;tag=as0c8c671a
Call-ID: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c418a8b"
Content-Length: 0
11:58:38.593[f74]SIP::Tx:ACK sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5061;received=10.10.10.157;rport=5061;branch=z9hG4bK00b777c9
From: 229 <sip:300@10.10.10.155>;tag=b7d4f3
To: <sip:260@10.10.10.155>;tag=as0c8c671a
Call-Id: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Content-Length: 0
11:58:38.609[f74]SIP::Tx:INVITE sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5061;rport;branch=z9hG4bK00b777ca
From: 229 <sip:300@10.10.10.155>;tag=b7d4f4
To: <sip:260@10.10.10.155>
Contact: <sip:300@10.10.10.157:5061>
Call-Id: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 INVITE
Max-Forwards: 70
Organization: 2E6A60B4-035E-4926-BA71-733FBF180176
Proxy-Authorization: Digest algorithm=md5,nonce="1c418a8b",realm="asterisk",
response="663ec6396a633d158d1109d9e14f9c98",uri="sip:260@10.10.10.155",username="300"
Content-Length: 222
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 12018140 12018140 IN IP4 10.10.10.157
s=LanScape
c=IN IP4 10.10.10.157
t=0 0
m=audio 8006 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
11:58:43.609[f74]SIP::Tx:CANCEL sip:260@10.10.10.155 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.157:5061;rport;branch=z9hG4bK00b777ca
From: 229 <sip:300@10.10.10.155>;tag=b7d4f4
To: <sip:260@10.10.10.155>
Call-Id: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScapeCorp.com)
Content-Length: 0
11:58:43.609[eb8]SIP::Rx:SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5061;branch=z9hG4bK00b777c9;received=10.10.10.157;rport=5061
From: 229 <sip:300@10.10.10.155>;tag=b7d4f3
To: <sip:260@10.10.10.155>;tag=as0c8c671a
Call-ID: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
11:58:43.609[eb8]SIP::Rx:SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.157:5061;branch=z9hG4bK00b777ca;received=10.10.10.157;rport=5061
From: 229 <sip:300@10.10.10.155>;tag=b7d4f4
To: <sip:260@10.10.10.155>;tag=as0c8c671a
Call-ID: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:260@10.10.10.155>
Content-Length: 0
11:58:44.625[f14]Channel::OnHook TerminateCall Error : SipCallTimeOut
11:58:44.953[eb8]SIP::Rx:SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5061;branch=z9hG4bK00b777c9;received=10.10.10.157;rport=5061
From: 229 <sip:300@10.10.10.155>;tag=b7d4f3
To: <sip:260@10.10.10.155>;tag=as0c8c671a
Call-ID: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
11:58:45.625[f74]SIP::Phone Line Notification:SipCallCanceled
11:58:45.625[f74]SIP::Phone Line Notification:SipOnHook
11:58:46.281[eb8]SIP::Rx:SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5061;branch=z9hG4bK00b777c9;received=10.10.10.157;rport=5061
From: 229 <sip:300@10.10.10.155>;tag=b7d4f3
To: <sip:260@10.10.10.155>;tag=as0c8c671a
Call-ID: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
11:58:48.937[eb8]SIP::Rx:SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.157:5061;branch=z9hG4bK00b777c9;received=10.10.10.157;rport=5061
From: 229 <sip:300@10.10.10.155>;tag=b7d4f3
To: <sip:260@10.10.10.155>;tag=as0c8c671a
Call-ID: 4e93a629-b543-472f-acc8-924d59098998-00000998@10.10.10.157
CSeq: 12023489 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
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If you see the bold items, you will see that "Rx:SIP/2.0 487 Request Terminated" is received in the same time as "Tx:CANCEL" is sent. Note that these messages are in time order and Cancel is sent first. Again despite I have set TerminateCall timeout to 1000 ms "Phone Line Notification:SipCallCanceled" event is received 2 seconds after TerminateCall was called. So I think this is the 2000 ms timeout which you mentioned in previous post. So I am sure increasing the timeout value will not solve any problem.
This situation occures in our lab about 1 time in each 5 outbound call. So testing this state is so easy for me. If you want you can send me one new release for test.
Regards,
Jalal Abedinejad
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 07 2008 at 12:19pm | IP Logged
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Thanks Jalal for the added information. We will be working on this issue after we complete some current work.
We will get in contact via email to access your Asterisk box if needed.
As always, your knowledge and effort is appreciated.
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 12 2008 at 11:35am | IP Logged
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Hi Jalal,
We will be looking into this right away tomorrow. If your offer is still available, we would like to access your v1.4.17-1 Asterisk box to test against. We look forward to locating this issue as quickly as possible.
What we need from you:
1) Two SIP extension numbers for test.
2) SIP authentication credentials for the above 2 user accounts.
If you can, send the above info to our support group email.
Have a good day and we will speak with you soon.
Support
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 13 2008 at 1:59am | IP Logged
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Hi,
Our VPN Connection info is sent to support group mail.
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 13 2008 at 10:07am | IP Logged
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Hi Jalal,
We will start testing right now. We can connect to your Asterisk box and register.
We will update you as we progress.
Thanks,
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 13 2008 at 5:56pm | IP Logged
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Jalal,
We have used your Asterisk box all day today. Thanks for the access. We will probably want to test a bit more tomorrow if you don't mind. If this is not possible, let us know. Otherwise we will assume that in the morning we will still have access to the three extensions you allowed us to use.
Best,
Support
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 13 2008 at 11:36pm | IP Logged
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No problem, but I want to know if you could see the problem or not?
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 16 2008 at 6:01pm | IP Logged
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Hi Jalal,
Thanks for the access to your VPN and your Asterisk box. Yes we did see the issue you reported.
We located a state issue in the media engine that should take care of the final INVITE ACK issue associated with CANCELed calls you reported. However, you test as required and report your results back to us.
We also added absolute SIP Rx/Tx logging time stamps accurate to +/- 1Ms. This should help us all moving forward. One thing in your Tazarv SIP logs, be aware that your absolute time stamps look like they have 15Ms resolution. You probably are already aware of this fact but just in case, we thought we would mention it.
We have an updated product image you can use temporarily for your VOIP development and testing. This temp image is actually a “sneak preview” of the v6 offering soon to be released. You can use this ‘DLL only” product image with your current installation by simply replacing your existing files with the new ones in the image and using your existing media engine license files.
You can download the temp development image from your support FTP account. See the “DLL Only v5.12.8.3” directory of your support FTP account. Note that this distribution only contains minimal content so that you can preview the new changes and continue with your development.
Please read the release notes that accompany the product image for the latest changes and news.
If you want to enable new v6 integrated DTMF related functionality, you will have to use the temp license that comes with the image. The trial license is good for a couple of months. If you eventually want to enable the new integrated DTMF capabilities (fully integrated in-band and RFC2833 DTMF generation and detection) you will have to purchase an upgrade to v6 sometime later this year at the current 4 line, single instance price. Otherwise we will be able to issue you a new v5 product license at no cost. In this case the v5 product license will not have internal DTMF capabilities enabled. We will probably consider this the last free upgrade we can offer you. Need to keep the bills paid :)
Repost with any questions or feedback.
Best regards,
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 16 2008 at 6:21pm | IP Logged
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Oh, two additional items….
We tested the media engine here with Asterisk v1.4.18.1.
1)
One thing we still see is if calls are very quickly CANCELed by the media engine before Asterisk gets a chance to send out its “407 Proxy Authentication Required” message for the first INVITE, Asterisk will never be satisfied no matter how many final INVITE ACKs we send. We did not look into the source code for Asterisk so we do not know why he is not being satisfied for this case. Anyway, below is a SIP log showing the behavior. Maybe you can spot the issue.
2)
We noticed this with your Asterisk box: Why does the Asterisk server seem to return so many 503 server error responses in between calls when hammering on it? (SIP/2.0 503 Server error).
Code:
Asterisk repeatedly sendning "407 Proxy Authentication Required" needlessly after
sending INVITE and 128Ms later CANCEL.
Its as if Asterisk is ignoring the INVITE ACK for the original 407.
************* Log Opened (May 16 18:20:37) *************
>>>> TxTxTxTxTx (#24, [18:20:43.271] 3446 Ms, To: 10.10.10.155:5060) >>>>
INVITE sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;rport;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>
Contact: <sip:220@10.10.10.9:5082>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 218
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=220 40423843 40423843 IN IP4 10.10.10.9
s=LanScape
c=IN IP4 10.10.10.9
t=0 0
m=audio 8008 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16
>>>> TxTxTxTxTx (#25, [18:20:43.399] 128 Ms, To: 10.10.10.155:5060) >>>>
CANCEL sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;rport;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#23, [18:20:43.639] 3791 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40082f04"
Content-Length: 0
>>>> TxTxTxTxTx (#26, [18:20:43.642] 243 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#24, [18:20:43.664] 25 Ms, From: 10.10.10.155:5060) <<<<
OPTIONS sip:220@10.10.10.9:5082 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.155:5060;branch=z9hG4bK45c4f9b4;rport
From: "Unknown" <sip:Unknown@10.10.10.155>;tag=as5dbccccd
To: <sip:220@10.10.10.9:5082>
Contact: <sip:Unknown@10.10.10.155>
Call-ID: 3eff6401166588f24e3f86301d82d42d@10.10.10.155
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Mar 2008 16:59:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
>>>> TxTxTxTxTx (#27, [18:20:43.667] 25 Ms, To: 10.10.10.155:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.155:5060;rport;branch=z9hG4bK45c4f9b4
From: "Unknown" <sip:Unknown@10.10.10.155>;tag=as5dbccccd
To: <sip:220@10.10.10.9:5082>;tag=16530300
Call-Id: 3eff6401166588f24e3f86301d82d42d@10.10.10.155
CSeq: 102 OPTIONS
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRx (#25, [18:20:43.750] 86 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
>>>> TxTxTxTxTx (#28, [18:20:43.754] 87 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#26, [18:20:43.774] 24 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:260@10.10.10.155>
Content-Length: 0
<<<< RxRxRxRxRx (#27, [18:20:44.843] 1069 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40082f04"
Content-Length: 0
>>>> TxTxTxTxTx (#29, [18:20:44.845] 1091 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRx (#28, [18:20:46.076] 1233 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40082f04"
Content-Length: 0
>>>> TxTxTxTxTx (#30, [18:20:46.079] 1234 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRx (#29, [18:20:48.551] 2475 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40082f04"
Content-Length: 0
>>>> TxTxTxTxTx (#31, [18:20:48.554] 2475 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRx (#30, [18:20:53.493] 4942 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40082f04"
Content-Length: 0
>>>> TxTxTxTxTx (#32, [18:20:53.496] 4942 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRx (#31, [18:20:58.465] 4972 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40082f04"
Content-Length: 0
>>>> TxTxTxTxTx (#33, [18:20:58.468] 4972 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRx (#32, [18:21:03.416] 4951 Ms, From: 10.10.10.155:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.9:5082;branch=z9hG4bK02694834;received=10.10.10.9;rport=5082
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-ID: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40082f04"
Content-Length: 0
>>>> TxTxTxTxTx (#34, [18:21:03.418] 4950 Ms, To: 10.10.10.155:5060) >>>>
ACK sip:260@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5082;received=10.10.10.9;rport=5082;branch=z9hG4bK02694834
From: 220 <sip:220@asterisk>;tag=2691301;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:260@asterisk>;tag=as1e1f1618
Call-Id: 0e589028-f6f5-4ef0-9790-ffae87845089-000030dc@10.10.10.9
CSeq: 6873740 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.3 (www.LanScapeCorp.com)
x-CustomHeader-Extension-220: "This is a modified transmitted SIP message."
Content-Length: 0
************* Log Closed (May 16 18:21:27) *************
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Thanks,
Support
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 19 2008 at 6:36am | IP Logged
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Hi,
Thanks for this new version. I tested this issue with our asterisk and it seems this problem does not occure anymore. I'm interested to know what was the problem in LMEVoip that cause this issue. In v5.12.8.1 sometimes the asterisk did not response to second invite. Was it related to this?
About 15 ms resolution of windows time functions you are correct. I did know about this but I did not care about it. Do you have any solution to log the time in 1 ms or less resolution?
Something that made me confused about using this new version 5.12.8.3 was removing GetMixerResampleState undocumented function. I know you are not responsible for undocumented functions, but I am telling to other customers that may read this topic that this version does not export GetMixerResampleState and SetMixerResampleState undocumented functions. My program was calling GetMixerResampleState just to log the state and as it was NULL in my program caused my program to be terminated abnormally.
About asterisk behaviour when Cancelling before Proxy Authentication I don't know what is the problem. I will search about this later.
About asterisk 503 server not found maybe it is because the low speed VPN connection or VMWare installation of the asterisk in our lab. Anyway I have not seen this problem in our lab.
Another Question: What features are you going to add to your v6 except RFC2833 support? Is there any other valuable feature?
Regards,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 19 2008 at 2:42pm | IP Logged
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Hi Jalal,
<<< You
I'm interested to know what was the problem in LMEVoip that cause this issue. In v5.12.8.1 sometimes the asterisk did not response to second invite. Was it related to this?
Support >>>
That’s the strange thing. When testing with our v1.4.18.1 Asterisk box and your 1.4.17-1 Asterisk box and VOIP Media Engine v5.12.8.3, we never saw Asterisk “not respond” to a transmitted INVITE request from the media engine.
What we did detect was: For calls that were CANCELed very fast (before Asterisk could send a 407 or 18x provisional response, the media engine would go to the cancel processing state in the internal call state machine. Simply stated: a call CANCEL and a 407 proxy auth response was overlapping. When the 407 finally arrived, it was possible for the media engine to simply dump the 407 and not final INVITE ACK back. This lead to Asterisk continually sending additional 407’s until the timeout period was reached by Asterisk.
<<< You
About 15 ms resolution of windows time functions you are correct. I did know about this but I did not care about it. Do you have any solution to log the time in 1 ms or less resolution?
Support >>>
We suspected you already knew. The 15 Ms resolution bugged us though. It makes too many things look like they are instantaneous when they are not. What we did was implement internal logic in one of our internal background threads that obtains the time of day (accurate to 10Ms or 15Ms depending on the OS and underlying hardware. We then used a technique to interpolate between the resolution of the time of day values using performance counter support every few seconds. This way we can re-synch time of day and performance counting to +/- 1Ms. Too bad the OS does not natively support this. Maybe we will offer this Ms accurate time of day to the API for end users in a future release. Here is an MSDN link that explains some of this:
Implement a Continuously Updating, High-Resolution Time Provider for Windows
Regarding GetMixerResampleState() and SetMixerResampleState() API procs:
Ohhhhh…. the export signatures for those two undocumented APIs were changed in v5.12.8.3. That’s the danger when we all use undocumented APIs. We most often forget to tell customers when they get changed or removed.
Here are the new prototypes:
Code:
extern "C"
{
TELEPHONY_RETURN_VALUE VOIP_API SetMixerResampleState(
SIPHANDLE hStateMachine,
BOOL MixerResampleEnableState,
int MixerResampleBlockTrigger,
BOOL MasterPlaybackResampleEnableState,
int MasterPlaybackResampleBlockTrigger
);
TELEPHONY_RETURN_VALUE VOIP_API GetMixerResampleState(
SIPHANDLE hStateMachine,
BOOL *pMixerResampleEnableState,
int *pMixerResampleBlockTrigger,
BOOL *pMasterPlaybackResampleEnableState,
int *pMasterPlaybackResampleBlockTrigger
);
}
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Before v5.12.8.3, the two enable states in the procs were the same. Now they are split. The “trigger” values before were not broken out to the API and internally defaulted to 3.
<<< You
About asterisk behaviour when Cancelling before Proxy Authentication I don't know what is the problem. I will search about this later.
Support >>>
OK.
<<< You
About asterisk 503 server not found maybe it is because the low speed VPN connection or VMWare installation of the asterisk in our lab. Anyway I have not seen this problem in our lab.
Support >>>
It looks like when we call the same extension immediately after terminating a call to that extension, we get the 503s. As you said, probably no issue here.
<<< You
Another Question: What features are you going to add to your v6 except RFC2833 support? Is there any other valuable feature?
Support >>>
The big thing is fully integrated in-band and RFC2833 DTMF support. Both generation and detection. The other big area that will be affected by the v6 release is conferencing/line bridging capability. Enhancements to the speed and efficiency at which lines can enter and leave conference sessions has been improved greatly. There are a lot of other general updates and fixes too. Some changes having to do with improving high line density server deployments, etc. Some of the changes you may have in your v5.12.8.1 image and some you do not. We will post a complete v6 change list at the web site soon.
We are also going to adopt a new manner in which we report version changes to the public. Some sort of running web based change log. It will make it much easier for end user to determine if an update is work the price or is applicable for their particular situation.
Thanks Jalal,
Support
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: May 20 2008 at 3:58am | IP Logged
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<<< Me
I'm interested to know what was the problem in LMEVoip that cause this issue. In v5.12.8.1 sometimes the asterisk did not response to second invite. Was it related to this?
Support >>>
That's the strange thing. When testing with our v1.4.18.1 Asterisk box and your 1.4.17-1 Asterisk box and VOIP Media Engine v5.12.8.3, we never saw Asterisk not respond to a transmitted INVITE request from the media engine.
What we did detect was: For calls that were CANCELed very fast (before Asterisk could send a 407 or 18x provisional response, the media engine would go to the cancel processing state in the internal call state machine. Simply stated: a call CANCEL and a 407 proxy auth response was overlapping. When the 407 finally arrived, it was possible for the media engine to simply dump the 407 and not final INVITE ACK back. This lead to Asterisk continually sending additional 407s until the timeout period was reached by Asterisk.
<<< Me
I was interested to know about the problem not sending Ack by LMEVoip for 487 Request Terminated. I think you have described another issue.
Thanks for other info,
Regards,
Jalal
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: December 31 2008 at 8:44am | IP Logged
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Hi Support,
Happy new year,
We had completely forgotten about v5.12.8.3 you had provided for us, has a time limitation. So we were using this version in our product, until we got "This product is expired." message from LMEVoip.dll today (2008/12/31) .
As you had mentioned in one of your previous posts on this topic, you would provide us "a new v5 product license at no cost" to work with this version. As our current license which worked with v5.12.8.3 is expired, please provide us a new "NOT TIME LIMITED" license.
Your sentence in one of previous posts:
Code:
Otherwise we will be able to issue you a new v5 product license at
no cost. In this case the v5 product license will not have internal
DTMF capabilities enabled. We will probably consider this the last free
upgrade we can offer you.
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We would upgrade our license later to v6.
Thanks again,
Jalal Abedinejad
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: December 31 2008 at 10:23am | IP Logged
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Hi Jalal,
Thanks for your post. I have been wondering what your group has been up to.
The last v5 product we officially gave you as v5.12.8.1. If you are using v5.12.8.3 engineering release and it has timed out, then you will have to upgrade to Release 6 media engine now not later. We are only going to support Release 6 code base moving forward. The engineering release is not meant for general deployment to your customers.
As a matter of fact, licensing terms have also changed. If you want to licese the Release 6 media engine in your product and have unlimited deployment capabilities, then you will have to relicense each year. There are no other options.
Thanks,
Randal
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: January 01 2009 at 1:47am | IP Logged
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Hi Randal,
When you had promissed us a new v5 product license at no cost, you had provided us v5.12.8.3 which had fixed some bugs for us. So I think you should give us the license to work with this version without any time-bomb limitation. Unless you dont't want to keep to your promise.
Unfortunately I should inform you if you have changed license terms to limit license for a year, we will never upgrade to v6 because we always contract with customers to provide the products without any time limitation, so if they can not use the product after one year, they will kill us ;) .
Regards,
Jalal
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: SIP 487 Request Terminated
