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juice Vetran
Joined: December 05 2006 Location: United States Posts: 139
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Posted: January 23 2007 at 6:31pm | IP Logged
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Hi,
I was trying the MultiLine Phone example project, everything was fine (initialization, proxy registration) up to the moment I want to place calls.
It was crashing immediately on each incoming call with following exception:
Unhandled exception at 0x7c812a5b in MultiLine Phone.exe: Microsoft C++ exception: SdpExceptions at memory location 0x0313c85c..
or on outgoing call, when called party has picked up, then exception was like this:
Unhandled exception at 0x7c812a5b in MultiLine Phone.exe: Microsoft C++ exception: SdpExceptions at memory location 0x0333c824..
bellow you can see the Sip Log of incoming call crash
************* Log Opened (Jan 23 15:54:13) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 83.0.0.1:5060) >>>>
REGISTER sip:voip.domain.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.128:5060;rport;branch=z9hG4bK014001e9
From: <sip:user01@voip.domain.com>;tag=14012be
To: <sip:user01@voip.domain.com>
Call-Id: 5e3da491-3550-4427-acfa-22fe2affa641-00000ec0@10.0.0.128
CSeq: 4173544 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:user01@10.0.0.128:5060>
User-Agent: LanScape VOIP Media Engine/5.12.3.10 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 83.0.0.1:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bK014001e9
From: <sip:user01@voip.domain.com>;tag=14012be
To: <sip:user01@voip.domain.com>
CSeq: 4173544 REGISTER
Call-ID: 5e3da491-3550-4427-acfa-22fe2affa641-00000ec0@10.0.0.128
Expires: 31
WWW-Authenticate: Digest realm="domain.com", algorithm=MD5, nonce="1105161e192", qop="auth", opaque="41bd561105161e192"
Server: SIP Proxy (1.0 SIP/2.0)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (340 Ms, To: 83.0.0.1:5060) >>>>
REGISTER sip:voip.domain.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.128:5060;rport;branch=z9hG4bK01400a01
From: <sip:user01@voip.domain.com>;tag=1400206
To: <sip:user01@voip.domain.com>
Call-Id: 5e3da491-3550-4427-acfa-22fe2affa641-00000ec0@10.0.0.128
CSeq: 4173545 REGISTER
Authorization: Digest algorithm=md5, cnonce="003fe98d", nc="00000001", nonce="1105161e192", opaque="41bd561105161e192", qop="auth", realm="domain.com", response="0072205d6fdd233fd41d86cda9491894", uri="sip:voip.domain.com", username="user01"
Expires: 3600
Max-Forwards: 70
Contact: <sip:user01@10.0.0.128:5060>
User-Agent: LanScape VOIP Media Engine/5.12.3.10 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (291 Ms, From: 83.0.0.1:5060) <<<<
SIP/2.0 200 OK Authenticated
Via: SIP/2.0/UDP 10.0.0.128;branch=z9hG4bK01400a01
From: <sip:user01@voip.domain.com>;tag=1400206
To: <sip:user01@voip.domain.com>
CSeq: 4173545 REGISTER
Call-ID: 5e3da491-3550-4427-acfa-22fe2affa641-00000ec0@10.0.0.128
Expires: 3785
Server: SIP Proxy (1.0 SIP/2.0)
Content-Length: 0
Contact: <sip:user01@voip.domain.com:5060>
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (19949 Ms, From: 83.0.0.1:5060) <<<<
INVITE sip:user01@83.0.0.7:5060 SIP/2.0
Via: SIP/2.0/UDP 83.0.0.1;branch=z9hG4bK551adcb492b7d85a638704728a5d256a
Via: SIP/2.0/UDP 10.0.0.128:53374;received=83.0.0.7;rport=53374;branch=z9hG4b K-d87543-5049d555270eeb0f-1--d87543-
Max-Forwards: 70
To: <sip:user01@voip.domain.com>
From: user02 <sip:user02@voip.domain.com>;tag=3e31024c
Call-ID: N2MzYzM0N2JkMmIwZDdmMDY0NTgzMmFmNWY1ZDc0YTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 257
Contact: <sip:user02@83.0.0.7:53374;maddr=83.0.0.1>
Record-Route: <sip:user01@83.0.0.1:5060>
v=0
o=- 2 2 IN IP4 83.0.0.12
s=CounterPath X-Lite 3.0
c=IN IP4 83.0.0.12
t=0 0
m=audio 20090 RTP/AVP 0 3 101
c=IN IPV4 83.0.0.12
a=alt:1 1 : 0qo9Sxb5 FSQeFa63 10.0.0.128 58864
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
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Jalal Vetran
Joined: April 24 2006 Location: Iran Posts: 188
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Posted: January 24 2007 at 2:32am | IP Logged
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Hi juice,support
After encountering the same problem as above in Single Line Phone (when ignoreing the call), and checking the source code we found the problem.
By default we thought the exe files located in Bin\Release folder are the latest compilation of the software examples. But it seems these examples are not compiled for Ver 5.12.3.10 of Lanscape. After rebuilding the projects the problem solved.
You know there are some functions changed in v5.12.3.10 such as AbortIncomingCall. So the samples should be recompiled to work with new version.
To Support:
Please put the latest build of your software examples in Bin\Release.
To Juice:
Recompile the software example of MultiLine Phone. I have not tested, but I think it will be solved.
Thanks
Jalal Abedinejad
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: January 24 2007 at 7:47am | IP Logged
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Hi juice,
I think Jalal is correct. Try rebuilding all of the examples. If that does not clear up the problem, repost to this thread.
Thanks Jalal for your input!
Support
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juice Vetran
Joined: December 05 2006 Location: United States Posts: 139
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Posted: January 24 2007 at 2:33pm | IP Logged
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first thank for your hints. Exceptions i listed above I get also with my application which is not based on example, I did refer to Multiphone only to be sure that this issue is not releated to my development, but anyway to give a try i have rebuild Multiphone and the result was the same.
So resuming, i am getting this exception on 2 independent applications.
Andrew
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: January 24 2007 at 4:06pm | IP Logged
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Andrew,
Ok, we are looking now...
Support
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: January 24 2007 at 4:13pm | IP Logged
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Andrew,
We took the INVITE you posted and manually sent it to a test media engine app we have here.
Sure enough, the media engine pukes on the INVITE from the X-Lite phone.
We will fix immediately. Please wait for us to repost to this thread.
Good work. Thanks for locating this bug.
Support
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: January 24 2007 at 5:08pm | IP Logged
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Andrew,
OK. We tracked down the problem.
First, a few comments on the INVITE the X-Lite soft phone is generating:
Item 1:
The SIP “Content-Length:” header is incorrect. It states that there are 257 bytes of SDP. In actuality there are only 246 bytes of SDP data in the INVITE. This is a CounterPath bug. The media engine detected this and defaulted to parsing the SDP anyway.
The content header should not be:
Content-Length: 257
but should be:
Content-Length: 246
Item 2: Multiple connection (c=) headers in the SDP. Why???
We do not know why there are multiple “c=” headers in the SDP. You will have to ask CounterPath about this. The Media Engine parsed the first one OK but when it got to the second one, it puked on the “IPV4” token. We agree, the media engine should have handled this better instead of blowing up.
As far as we can tell, the connection header must specify either IP4 or IP6. We looked in RFC 2327 “SDP: Session Description Protocol” and we do not see any mention regarding the “address type” (addrtype) type denoted as “IPV4”. See “Appendix A: SDP Grammar” of the RFC. You will have to ask CounterPath about this also. Why are they using IPV4? Who knows.
We added the appropriate changes to the media engine to handle this “IPV4” situation and it looks OK now. Please request a new trial product image so you can continue your testing.
Thanks again for your input. Interoperability issues are inevitable.
Repost as needed,
Support
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support Administrator
Joined: January 26 2005 Location: United States Posts: 1666
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Posted: January 24 2007 at 5:21pm | IP Logged
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Andrew,
Here is a SIP log of the media engine handling your INVITE:
Code:
************* Log Opened (Jan 24 17:19:22) *************
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (41812 Ms, From: 192.168.1.2:1366) <<<<
INVITE sip:user01@83.0.0.7:5060 SIP/2.0
Via: SIP/2.0/UDP 83.0.0.1;branch=z9hG4bK551adcb492b7d85a638704728a5d256a
Via: SIP/2.0/UDP 10.0.0.128:53374;received=83.0.0.7;rport=53374;branch=z9hG4b K-d87543-5049d555270eeb0f-1--d87543-
Max-Forwards: 70
To: <sip:user01@voip.domain.com>
From: user02 <sip:user02@voip.domain.com>;tag=3e31024c
Call-ID: N2MzYzM0N2JkMmIwZDdmMDY0NTgzMmFmNWY1ZDc0YTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 257
Contact: <sip:user02@83.0.0.7:53374;maddr=83.0.0.1>
Record-Route: <sip:user01@83.0.0.1:5060>
v=0
o=- 2 2 IN IP4 83.0.0.12
s=CounterPath X-Lite 3.0
c=IN IP4 83.0.0.12
t=0 0
m=audio 20090 RTP/AVP 0 3 101
c=IN IPV4 83.0.0.12
a=alt:1 1 : 0qo9Sxb5 FSQeFa63 10.0.0.128 58864
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (13375 Ms, To: 83.0.0.7:53374) >>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.0.0.1:5060;received=192.168.1.2;branch=z9hG4bK551adcb492b 7d85a638704728a5d256a
Via: SIP/2.0/UDP 10.0.0.128:53374;received=83.0.0.7;rport=53374;branch=z9hG4b K-d87543-5049d555270eeb0f-1--d87543-
x-MyCustomHeader: "This is a modified transmitted SIP message."
From: user02 <sip:user02@voip.domain.com>;tag=3e31024c
To: <sip:user01@voip.domain.com>
Call-ID: N2MzYzM0N2JkMmIwZDdmMDY0NTgzMmFmNWY1ZDc0YTM.
CSeq: 2 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.3.13 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 83.0.0.7:53374) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 83.0.0.1:5060;received=192.168.1.2;branch=z9hG4bK551adcb492b 7d85a638704728a5d256a
Via: SIP/2.0/UDP 10.0.0.128:53374;received=83.0.0.7;rport=53374;branch=z9hG4b K-d87543-5049d555270eeb0f-1--d87543-
x-MyCustomHeader: "This is a modified transmitted SIP message."
From: user02 <sip:user02@voip.domain.com>;tag=3e31024c
To: <sip:user01@voip.domain.com>;tag=25ecd4d
Call-ID: N2MzYzM0N2JkMmIwZDdmMDY0NTgzMmFmNWY1ZDc0YTM.
CSeq: 2 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.3.13 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (3547 Ms, To: 83.0.0.7:53374) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.0.0.1:5060;received=192.168.1.2;branch=z9hG4bK551adcb492b 7d85a638704728a5d256a
Via: SIP/2.0/UDP 10.0.0.128:53374;received=83.0.0.7;rport=53374;branch=z9hG4b K-d87543-5049d555270eeb0f-1--d87543-
Record-Route: <sip:user01@83.0.0.1>
From: user02 <sip:user02@voip.domain.com>;tag=3e31024c
To: <sip:user01@voip.domain.com>;tag=25ecd4d
Call-Id: N2MzYzM0N2JkMmIwZDdmMDY0NTgzMmFmNWY1ZDc0YTM.
CSeq: 2 INVITE
Contact: <sip:3-3_3.3!3~3*3@192.168.1.2:6060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
User-Agent: LanScape VOIP Media Engine/5.12.3.13 (www.LanScapeCorp.com)
Content-Length: 172
Content-Type: application/sdp
v=0
o=LanScape 3378669573 3378669573 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 28154 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
************* Log Closed (Jan 24 17:19:39) *************
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