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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 06 2009 at 5:15pm | IP Logged
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We have a VOIP application that places outbound calls and plays a sequence of wave files to the listener.
We have an issue with our outgoing phone calls, they are automatically hanging up after about 30 seconds. We noticed that we are receiving a SipByeReceived from our carrier after about 30 seconds into the call. This happens every single time and is very predictable.
We then started using the Media Engine EnableKeepAliveTransmissions method thinking that our carrier was dropping the call because of inactivity. The EnableKeepAliveTransmissions did not help.
So we contacted our carrier and this is what they said:
I believe the ACK your PBX sends us for the 200 ok is the problem. We time out the call because we do not receive a proper ACK.
Here is the ACK your PBX sends.
Note: IP addresses changed for security.
Code:
ACK sip:+13054969343@1.2.3.4;vsf=AAAAAAAAAAAAAAAAAAAAAAAA A AAAAAAAAAAAAAA7b3RnPXB2cA-- SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;received=5.6.7.8;rport=5060;b ranch=z9hG 4bK064a9de3
From:"Techcierge OnCall";tag=64a8686
To:;tag=gK0eaaaeb6
Call-ID: 5f286f8a-1353-486d-b925-270315f19c52-00001044@5.6.7 .8
CSeq: 4903076 ACK
Max-Forwards: 70
Route:,
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScape Corp.com)
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScape Corp.com)
Content-Length: 0
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One thing that seems odd is the CSeq: 4903076 ACK.
This is normally set to CSeq: 102 ACK
But the one big issue I feel that needs attention is Route:,. A good example is Route:. I do not believe the comma is supposed to be in the route and also port 5060 should be right behind the ip separated by a semi colon. It may be best to check with your PBX vendor on this.
What are yout thoughts? How can we resolve this issue?
Thanks.
-AJ
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 06 2009 at 5:24pm | IP Logged
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I logged our SIP messages using LogPhoneLineSipMessages and this is the result:
Note: IP addresses changed for security.
Code:
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#0, [17:28:40.278] 0 Ms, To: 1.2.3.4:5060) >>>>
INVITE sip:+13054969343@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;rport;branch=z9hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>
Contact: <sip:8662418523@192.168.1.10:5060>;x-inst="VGhpcyBpcyB hIEMjIGNhbGwgdGVzdC4="
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Max-Forwards: 70
Organization: 7B71036A-396F-4B3B-A5FB-3198ACD52908
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 231
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=8662418523 110173812 110173812 IN IP4 192.168.1.10
s=LanScape
c=IN IP4 192.168.1.10
t=0 0
m=audio 8014 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 (#0, [17:28:40.351] 0 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#1, [17:28:43.144] 2793 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#2, [17:28:52.755] 9611 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#1, [17:28:52.792] 12514 Ms, To: 1.2.3.4:5060) >>>>
ACK sip:+13054969343@1.2.3.4;vsf=AAAAAAAAAAAAAAAAAAAAAAAA AAAAAAAAAAAAAAA7b3RnPXB2cA-- SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 ACK
Max-Forwards: 70
Route: <sip:1.2.3.4;vsf>,<sip:+13054969343@67.231.0 .69>
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#3, [17:28:53.190] 435 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#4, [17:28:54.163] 973 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#5, [17:28:56.190] 2027 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#6, [17:29:00.176] 3986 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4;otg=pvp>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#7, [17:29:04.283] 4107 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4;otg=pvp>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#8, [17:29:08.212] 3929 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4;otg=pvp>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#9, [17:29:12.135] 3923 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;received=192.168.1.10;rport=5060;branch=z9 hG4bK0691563b
From: "Techcierge OnCall" <sip:8662418523@1.2.3.4;otg=pvp>;tag=6918796
To: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 9535113 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+13054969343@11.22.33.44:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,P RACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 15795 29877 IN IP4 11.22.33.44
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 58386 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#10, [17:29:16.255] 4120 Ms, From: 1.2.3.4:5060) <<<<
BYE sip:8662418523@192.168.1.10:5060 SIP/2.0
Record-Route: <sip:1.2.3.4;lr;ftag=gK0d8e9132>
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK367b.41a47966.0
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK0dBfe037570319d44e1
From: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
To: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 7251 BYE
Max-Forwards: 69
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#2, [17:29:16.271] 23479 Ms, To: 1.2.3.4:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.4:5060;received=1.2.3.4;branch=z9hG4bK36 7b.41a47966.0
Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK0dBfe037570319d44e1
Record-Route: <sip:1.2.3.4>
From: <sip:+13054969343@1.2.3.4>;tag=gK0d8e9132
To: "Techcierge OnCall" <sip:8662418523@1.2.3.4>;tag=6918796
Call-ID: cf7c7903-7dc5-4561-99ab-36d3c4cd376b-00001324@192.168.1.10
CSeq: 7251 BYE
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 0
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 06 2009 at 5:26pm | IP Logged
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BTW, the log above is without the Keep Alive enabled.
I also have an Ethereal Capture Dump in case this helps shed some light.
Thanks.
-AJ
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 06 2009 at 6:11pm | IP Logged
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Hi Aj,
Thanks for the post. Don’t worry, it will be something simple in the SIP that is causing the ACK to not be accepted by the service provider.
Item 1:
The first ACK you posted is completely hacked up. The “To:” header and the “Route:” headers are complete garbage. Where did you get this data? Are you saying the media engine transmitted this junk? This is very hard to believe. To put the ACK in context – I would require the complete SIP for the call setup. Looking only at the ACK will not show the entire picture or the possible snafu.
Also for the person at the service provider to make the statement regarding the CSeq number:
“This is normally set to CSeq: 102 ACK”
This statement makes no sense to me. The CSeq number will vary all over the map for different call dialogs (call legs).
Hmm… strange…... possibly a SIP noobie at the other end?
Item 2:
Thanks for posting your SIP log. Yes, it is evident that they are not processing the ACK we send. Hmm... I have to think…..
Do me a favor, email the exact same SIP log you posted to me using the LanScape support group email. The support forum may not preserve the posted SIP log perfectly and I need to see the unaltered “virgin” SIP for the call.
The ACK looks good. I wonder if their equipment is taking the ACK’s “Route:” header and passing the ACK on to the second endpoint specified in the route header. In other words, I bet their equipment is forwarding the ACK on to IP address 67.231.0.69 which may be the problem depending on how they are set up. This is only a guess.
Item 3:
It is the end of the day here. We probably will have to get back to this on Monday. If you can, is it possible for you to forward a test account to us for this service provider so that we can conduct testing from my location (Minneapolis)? I will also need a target number to call.
Item 4:
One thing you can do in the mean time is to hook the transmitted ACK messages being transmitted to the service provider and alter them in your app code. You can do this by processing the SipModifySipMessage immediate event for transmitted SIP messages.
I would start by removing the second route location in the ACK Route header and see what happens. You could also try other things that come to mind. Most times to find the SIP inter-op issue is a “matter of trial and error” and “process of elimination” regarding suspect SIP headers.
Thanks,
Randal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 10 2009 at 10:03am | IP Logged
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Hi Aj,
Any further word on this issue? I am on the road and will be back tomorrow. Did you try to narrow down the cause as I mentioned in item 4 above?
Thanks,
Randal
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 10 2009 at 11:00am | IP Logged
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Hi Randal,
The files you requested have been sent to you via email.
I sent your comments to our support rep at Bandwidth.com to see if he can find the problem, but have not heard back from him yet. I will follow up with him.
You mention a concern that their equipment might be "taking the ACK’s “Route:” header and passing the ACK on to the second endpoint specified in the route header. In other words, I bet their equipment is forwarding the ACK on to IP address 67.231.0.69 which may be the problem depending on how they are set up." I will mention that to them.
When we did enable KeepAliveTransmissions, we noticed a SIP message coming from 67.231.0.69 (their IP) saying "500 Internal Server Error". Not sure why?
I think a test on your end would be very beneficial. Our provider (Bandwidth.com) limits the IP addresses that can access their SIP servers. You would need to provide us your IP address, and then we could call them to add it to their white list. We would remove it after the test of course. The target number to call could be any number I suppose.
You suggested to modify the outgoing SIP message. How would we go about modifying the SIP message to alter the Route portion? (this is new to us). Why would it have two route location? Is this something that changed in the Media Engine since maybe version 5.12?
Thanks for all your assistance.
-AJ
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 11 2009 at 11:17am | IP Logged
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Hi Aj,
As I look at the ACK that is being sent, the request header:
Code:
ACK sip:+13054969343@216.82.224.202;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-- SIP/2.0
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…does not look correct. Not sure why the media engine is placing the “vsf=” parameter in the header. Possibly the results of a SIP inter-op “200 OK” parse issue. Notice the “vsf=” parameter in the “Record-Route:” header of the “200 OK” response.
Code:
Record-Route: <sip:1.2.3.4:5060;lr;ftag=6918796;vsf=AAAAAAAAAAAA AAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
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In the essence of speed, we should perform a test from my location. I will send you the LanScape IP address.
I will need from you the typical SIP user agent authentication information or whatever will allow me to place a few test calls through the VOIP provider. This issue will be simple to identify and resolve.
I will send you an email with my IP address. We will conduct test calls as soon as you email back the user agent config info.
Thanks,
Randal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 11 2009 at 11:29am | IP Logged
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Hi Aj,
I almost forgot to answer your other questions…
<<< You
You suggested to modify the outgoing SIP message. How would we go about modifying the SIP message to alter the Route portion? (this is new to us).
Support >>>
You can process the SipModifySipMessage event in your main media engine callback handler. When you process this message, your VOIP app can access either received SIP messages or “ready-to-be-transmitted” SIP messages. Your app can modify the SIP messages pretty much however you like.
<<< You
Why would it have two route location?
Support >>>
The first route location is to the service provider’s SIP proxy and the second is simply the location of the call endpoint as detected in the “200 OK” responses “Contact:” header. This is normal.
<<< You
Is this something that changed in the Media Engine since maybe version 5.12?
Support >>>
Nope.
Thanks,
Randal
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 11 2009 at 11:32am | IP Logged
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Hi Randal,
I received a voice mail from another one of their support reps today and this is what he said...
"On our 200 OK message we send to you, you should be pulling the IP from the Contact field (xx.xx.xx.69). But you are sending the ACK back directly to xxx.xx.xxx.202, our server doesn't know where to go to and sends another 200 over and over until it times out. In your ACK and your URI use the IP address in the Contact field we are sending you."
Hope this helps too. Thanks.
-AJ
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 11 2009 at 7:03pm | IP Logged
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Hi Alex,
Thanks for your VOIP service provider's config info. We have created a test setup here and see a similar issue that you reported. Perfect.
At this point we do not know where the issue lies (LanScape or the service provider) but seeing we are able to reproduce the situation, we can figure it our pretty quickly.
Here is a SIP log that represents initial test calls from our lab. We will continue testing immediately in the morning and repost to this thread when we discover the solution.
Good job so far, Thanks,
Randal
Note: IP addresses and destination phone number obfuscated for security purposes.
Code:
************* Log Opened (Feb 11 18:53:12) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#1, [18:53:12.984] 0 Ms, To: 1.2.3.4:5060) >>>>
REGISTER sip:1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5086;rport;branch=z9hG4bK01c4e8eb
From: <sip:8662418523@1.2.3.4>;tag=1c510e1
To: <sip:8662418523@1.2.3.4>
Call-ID: f98ac75a-24b8-4da6-94ed-71ed0c45bb13-000010c8@192.168.1.2
CSeq: 12909832 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:8662418523@192.168.1.2:5086>;user=phone
User-Agent: LanScape Utility Softphone/5.10.0.7
x-CustomHeader-Extension-8662418523: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.8 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#1, [18:53:13.046] 0 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;rport=5086;branch=z9hG4bK01c4e8eb;received=98.240.142.80
From: <sip:8662418523@1.2.3.4>;tag=1c510e1
To: <sip:8662418523@1.2.3.4>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.8560
Call-ID: f98ac75a-24b8-4da6-94ed-71ed0c45bb13-000010c8@192.168.1.2
CSeq: 12909832 REGISTER
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#2, [18:53:19.734] 6750 Ms, To: 1.2.3.4:5060) >>>>
INVITE sip:+19529438222@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5086;rport;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>
Contact: <sip:8662418523@192.168.1.2:5086>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-CustomHeader-Extension-8662418523: "Modified transmitted SIP message."
x-PhoneLine: 0
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.8 (www.LanScapeCorp.com)
Content-Length: 228
User-Agent: LanScape Utility Softphone/5.10.0.7
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=8662418523 29670718 29670718 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 22008 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 (#2, [18:53:19.796] 6750 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.1.2:5086;rport=5086;branch=z9hG4bK01c537a5;received=98.240.142.80
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#3, [18:53:21.437] 1641 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#4, [18:53:25.406] 3969 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#3, [18:53:25.406] 5672 Ms, To: 1.2.3.4:5060) >>>>
ACK sip:+19529438222@1.2.3.4;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-- SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 ACK
Max-Forwards: 70
Route: <sip:1.2.3.4;vsf>,<sip:+19529438222@67.231.0.69>
User-Agent: LanScape Utility Softphone/5.10.0.7
x-CustomHeader-Extension-8662418523: "Modified transmitted SIP message."
x-PhoneLine: 0
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.8 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#5, [18:53:25.890] 484 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#6, [18:53:26.906] 1016 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#7, [18:53:28.890] 1984 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#8, [18:53:32.890] 4000 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4;otg=pvp>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#9, [18:53:36.875] 3985 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4;otg=pvp>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#10, [18:53:40.875] 4000 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4;otg=pvp>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#11, [18:53:44.875] 4000 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c537a5
From: 8662418523 <sip:8662418523@1.2.3.4;otg=pvp>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898670 INVITE
Record-Route: <sip:1.2.3.4:5060;lr;ftag=1c4e252;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA7b3RnPXB2cA-->
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+19529438222@67.231.0.69:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 235
Content-Disposition: session; handling=optional
Content-Type: application/sdp
v=0
o=Sonus_UAC 13834 14040 IN IP4 67.231.0.69
s=SIP Media Capabilities
c=IN IP4 67.231.0.125
t=0 0
m=audio 37904 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:40
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#4, [18:53:54.125] 28719 Ms, To: 1.2.3.4:5060) >>>>
BYE sip:+19529438222@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5086;rport;branch=z9hG4bK01c583e1
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898671 BYE
Max-Forwards: 70
Route: <sip:1.2.3.4;vsf>,<sip:+19529438222@67.231.0.69>
User-Agent: LanScape Utility Softphone/5.10.0.7
x-CustomHeader-Extension-8662418523: "Modified transmitted SIP message."
x-PhoneLine: 0
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.8 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#12, [18:53:54.203] 9328 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 483 Something wrong in SIP Message - created loop
Via: SIP/2.0/UDP 192.168.1.2:5086;received=98.240.142.80;rport=5086;branch=z9hG4bK01c583e1
From: 8662418523 <sip:8662418523@1.2.3.4>;tag=1c4e252;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:+19529438222@1.2.3.4>;tag=gK0ea2a1c9
Call-ID: e2636b24-df41-422e-bc72-d5e125b72131-000010c8@192.168.1.2
CSeq: 12898671 BYE
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#5, [18:53:57.953] 3828 Ms, To: 1.2.3.4:5060) >>>>
REGISTER sip:1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5086;rport;branch=z9hG4bK01c59ab0
From: <sip:8662418523@1.2.3.4>;tag=1c552b5
To: <sip:8662418523@1.2.3.4>
Call-ID: f98ac75a-24b8-4da6-94ed-71ed0c45bb13-000010c8@192.168.1.2
CSeq: 12909833 REGISTER
Expires: 0
Max-Forwards: 70
Contact: <sip:8662418523@192.168.1.2:5086>;user=phone
User-Agent: LanScape Utility Softphone/5.10.0.7
x-CustomHeader-Extension-8662418523: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.8 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#13, [18:53:58.015] 3812 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5086;rport=5086;branch=z9hG4bK01c59ab0;received=98.240.142.80
From: <sip:8662418523@1.2.3.4>;tag=1c552b5
To: <sip:8662418523@1.2.3.4>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.c5c6
Call-ID: f98ac75a-24b8-4da6-94ed-71ed0c45bb13-000010c8@192.168.1.2
CSeq: 12909833 REGISTER
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
************* Log Closed (Feb 11 18:53:58) *************
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 13 2009 at 7:13am | IP Logged
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Hi Alex,
We worked on this yesterday and have not isolated the cause of the issue associated with the ACK that is being transmitted to the SIP server. We also tested with another SIP device from another vendor and had issues with one way audio. That being said, I know we will get to the bottom of the issue.
Item 1:
Initially I though the SIP server was some VOIP service provider your company uses but I began to realize that the SIP server may be some equipment that your company owns/controls. Is this correct?
Item 2:
Do you know the type/make/model of the SIP server? Sonus? If you do not want to expose this type of information to this support forum, send an email to my support group.
Item 3:
If this SIP server is not your equipment, do you have contact info for the service provider’s support? I may need to speak with them. Send this info via email instead of posting it to this support forum.
Item 4:
We have exhausted the support hours for February. Please speak with Matt to get an additional 16 hours of support work authorized. That should be enough to help us get this finished. We will simply invoice for the support after we have the solution and you test at your location.
Thanks Aj,
Randal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 16 2009 at 8:33pm | IP Logged
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Hi Aj,
The media engine has been updated to allow for a variable number of “Record-route:” and “Route:” header URI parameters. The Bandwidth.com service provider is using Sonus equipment that puts a variable number of parameters in the record route headers. If the variable parameters are not send back to the Sonus equipment in the final INVITE request’s ACK exactly like the were received in the “200 OK” response to the INVITE (in the Record-route headers), the Sonus equipment will ignore the ACK the media engine sends. The Sonus equipment will then send multiple “200 OK” responses. This eventually leads to the call timing out at their end which they subsequently terminate (in about 30 – 40 seconds).
If you want, you can look at the following SIP log that is in your support FTP account: “TechciergeSipLog 2-16-09.log”.
The SIP log shows us making a call using G729 and then terminating it.
We are not done yet…
We need to conduct further testing. There is still a one-way audio issue that we have detected with their service. We also want to test other media engine basics so that we catch everything now before we update your code. If you can, I need you to do a few things for me:
1)
Need 2nd WAN IP address enabled:
We want to perform both global IP and router/NAT testing. To do this, we need another IP address registered with the service provider. I will email the details to you. I called the service provider last week and the tech said they would enable the second IP I needed. Maybe they entered the wrong IP because using the 2nd WAN IP from here – I cannot register with their equipment. Our NAT router WAN IP is OK.
2)
Would like to test all of the media engine’s functionality with the service provider:
I want to test all the call functions of the media engine against their service (make calls, call holds, terminate calls etc). To do this, I need both IPs from item 1 above to be able to receive incoming calls from the PSTN side. Currently it looks like I can only test outgoing calls from the VOIP side. Let me now if this is possible. I would also like to be able to configure 2 VOIP numbers so that I can perform full VOIP to VOIP testing also.
I will send you an email containing some of the sensitive details right after I post this.
Thanks Aj,
Randal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 19 2009 at 1:25pm | IP Logged
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Hi Aj,
We updated the media engine and performed preliminary testing of the changes. The issue with the final ACK is resolved.
Please download the following ZIP archive from your support FTP account:
"VOIP Media Engine Engineering Release v6.0.0.8 Expires 3-31-09.zip"
The image is a 100% functional engineering release that expires the end of March. Please use this image to verify your call functionality.
Regarding deploying behind NAT/Firewalls:
Make sure your VOIP app calls the SetWanIpAddress() API procedure. Bandwidth.com's Sonus VOIP gateways will not handle the media engine calls properly if the SIP call setups use private LAN IP addresses.
From our testing here, everything looked ok and performed as expected.
Please repost with further issues and test results.
I will also send an email to Matt informing him of this post.
Thanks Aj,
Randal
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 19 2009 at 4:21pm | IP Logged
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Your message has been received. I will try this new DLL and let you know our test results.
Thanks for all your help.
-Alex
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 24 2009 at 1:02pm | IP Logged
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Hi Support,
Just FYI for you and other customers.
This new image (v6.0.0.8) was working on our dev environment, yet in production it was giving us the following errors in the event viewer:
---------------------------------------
*Dependent Assembly Microsoft.VC90.CRT could not be found and Last Error was The referenced assembly is not installed on your system.
*Resolve Partial Assembly failed for Microsoft.VC90.CRT. Reference error message: The referenced assembly is not installed on your system.
*Generate Activation Context failed for LMEVoipManaged.dll. Reference error message: The referenced assembly is not installed on your system.
---------------------------------------
After a lot of research I found out that this new image requires a very specific version of the Visual C++ libraries. It requires version 9.0.30729.1.
In our dev environment we had Microsoft VC90 CRT v9.0.30729.1 installed, it worked, but in our production machine we had a slightly older version 9.0.21022.8.
Obviously the previous LME image did not care. I believe this strict dependency is specified in the project's manifest. Most likely, this new version, in its project's manifest, where it describes its dependencies on Visual C++ libraries, it is specifying a very specific version (9.0.30729.1.). I think you can make it more generic by specifying a version number like this "9.0.xxxxx.y".
Again, just FYI. I hope it helps other customers.
This is the Microsoft article that helped me:
http://msdn.microsoft.com/en-us/library/ms235342.aspx
Thanks.
-Alex
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: February 24 2009 at 1:06pm | IP Logged
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BTW, preliminary test (very quick ones) resulted in not receiving the SipBye message and the call not hanging up after 30 Seconds.
We will be conducting more tests in the next few days. But so far, so good.
Thanks for all your help so far. We will let you know the results of the more extensive tests before we close this chapter and yell "Hurray!".
-Alex
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: February 24 2009 at 4:16pm | IP Logged
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Hi Aj,
Thanks for your post. Glad to know that your quick initial tests showed positive results.
I thought we had previously resolved the dependency issue. See this following post:
Could not load file or assembly ’LMEVoipManaged, Version=6.0:
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=557&PN =2
…and here:
Distributing Release 6 .NET VOIP applications:
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=546&PN =2
Unless I am missing something, we previously discussed that that media engine (native and managed code wrapper) requires the Microsoft Visual C++ 2008 SP1 Redistributable Package be installed on the target host computer. Maybe I am not remembering properly…..
Your suggestion is a good one but we have been burned before when manually overriding manifest settings. Anytime we override settings in the manifest, we have to keep track of compatibilities between different runtime lib versions ourselves which can lead to runtime problems down the road that are not easily tracked down. At least with the current mechanism, the binaries fail to load. When a load problem occurs, one must think immediately – “I have a dependency issue and I probably do not have some required software installed”.
I have no good answer. “DLL hell” from the past was never solved technically and all this manifest twiddling is just as bad. All we can do to keep things simple is to insist the runtime lib versions that were used to build the media engine product are installed on the target host machine.
Randal
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: March 24 2009 at 3:52pm | IP Logged
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Hi Support,
As you are aware, we download and installed the version you provided:
"VOIP Media Engine Engineering Release v6.0.0.8 Expires 3-31-09.zip"
This DLL expires 3/31. We need to get one that does not expire. We have an application in production that will need it.
Thanks,
-AJDiaz
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 25 2009 at 10:20am | IP Logged
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Hi Aj,
Yes, I know we need to update your code. I sent an email to Matt on Monday and have not gotten a response. We are looking to clear an invoice before we release the changes to you.
Thank you,
Randal
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: March 30 2009 at 2:26pm | IP Logged
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Hi Randal,
The check was mailed last week. Have you received it?
-AJD
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 30 2009 at 7:35pm | IP Logged
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Hi Aj,
Unfortunately we have not. :(
Randal
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: March 31 2009 at 2:49pm | IP Logged
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Randal,
It was just confirmed by our acct dept that the check was mailed last week on Thurs/Fri, so it should arrive very shortly.
Sorry if there is any delay. Please let us know when it does and please have our build ready to be sent. It would be greatly appreciated.
Thanks.
-AJD
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ajdiaz
Junior
Joined: December 10 2007
Location: United States
Posts: 76
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| Posted: April 03 2009 at 10:32am | IP Logged
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Randal,
Can you check on this status for me? The check was mailed last week so I'm thinking it should have arrived by now. Let me know. Thanks.
-AJD
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: April 03 2009 at 1:23pm | IP Logged
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Hello Aj,
The latest 8 line and 512 line VOIP Media Engine product images have been placed into your support FTP account for download.
In case you are not aware, all FTP access has been secured. Check out the following web page for further info:
Secure access to customer support FTP accounts:
http://www.lanscapecorp.com/support/SecureFtpAccess/FtpAcces sInstructions.asp
Repost if any issues.
Thank you,
Randal
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Receiving SipByeReceived After 30 Seconds Of Outbound Calls
