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PaulR
Intermediate
Joined: June 24 2011
Location: Philippines
Posts: 9
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| Posted: July 21 2011 at 12:52pm | IP Logged
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Hi,
I have question in Record Route and ModifySipMessage
1. Why does Lanscape sent back a differrent Record-Route compare to the original invite? See log below..
2. I tried to modified the transmitted SIP messages, but it does not work. I used the ModifySipMessage API to edit the sip message under SipModifySipMessage event but no success.
Code:
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#139, [09:59:52.042] 6513 Ms, From: 10.x.x.x:5060) <<<<
INVITE sip:3151@10.x.x.1:5060;x-sipX-nonat SIP/2.0
Record-Route: <sip:10.x.x.x:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2Aauth%7E.%2Afrom%7ERjIwN0tjdDgyQkRnSw%60%60.900_ntap%2Aid%7EMjU0MjAtMzQw%215cf18f2e8dcead823e8efb4f132bf3ff>
Via: SIP/2.0/UDP 10.x.x.x;branch=z9hG4bK-XX-13e44oV`61KkmxjGUEvk3KaZ2g
Via: SIP/2.0/UDP 10.x.x.x;branch=z9hG4bK-XX-13e04qcW67ka_ASCscr`7znLrQ~zQUUUs9llK2c3ejN5sFcTg;id=25420-340
Via: SIP/2.0/UDP 10.x.x.x:15060;rport=15060;branch=z9hG4bK1433Za0Q89gHH
Max-Forwards: 18
From: "2318232738*970742*112*9080" <sip:2318232738@10.x.x.x>;tag=F207Kct82BDgK
To: <sip:3151@ca1acd03.infonxx.local;sipx-noroute=VoiceMail;sipx-userforward=false>
Call-Id: c3a27f5f-2d83-122f-4398-005056ae006e
Cseq: 15250932 INVITE
Contact: <sip:mod_sofia@10.x.x.x:15060;x-sipX-nonat>
User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110326T074140Z
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 235
X-Dnis: 9080
X-Fs-Support: update_display
Remote-Party-Id: "2318232738*970742*112*9080" <sip:2318232738@10.x.x.x>;party=calling;screen=yes;privacy=off
Date: Wed, 20 Jul 2011 14:59:52 GMT
Expires: 20
X-Sipx-Handled: X10.112.75.13-10.x.x.x
v=0
o=FreeSWITCH 1311162907 1311162908 IN IP4 10.x.x.x
s=FreeSWITCH
c=IN IP4 10.x.x.x
t=0 0
m=audio 11074 RTP/AVP 9 0 8 98 10 101 13
a=rtpmap:98 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#54, [09:59:52.056] 28927 Ms, To: 10.x.x.x:5060) >>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.x.x.x:5060;received=10.x.x.x;branch=z9hG4bK-XX-13e44oV`61KkmxjGUEvk3KaZ2g
Via: SIP/2.0/UDP 10.x.x.x:5060;branch=z9hG4bK-XX-13e04qcW67ka_ASCscr`7znLrQ~zQUUUs9llK2c3ejN5sFcTg
Via: SIP/2.0/UDP 10.x.x.x:15060;rport=15060;branch=z9hG4bK1433Za0Q89gHH
From: "2318232738*970742*112*9080" <sip:2318232738@10.x.x.x>;tag=F207Kct82BDgK
To: <sip:3151@ca1acd03.infonxx.local;sipx-noroute=VoiceMail;sipx-userforward=false>
Call-Id: c3a27f5f-2d83-122f-4398-005056ae006e
Cseq: 15250932 INVITE
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#55, [09:59:52.061] 5 Ms, To: 10.x.x.x:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.x.x.x:5060;received=10.x.x.x;branch=z9hG4bK-XX-13e44oV`61KkmxjGUEvk3KaZ2g
Via: SIP/2.0/UDP 10.x.x.x:5060;branch=z9hG4bK-XX-13e04qcW67ka_ASCscr`7znLrQ~zQUUUs9llK2c3ejN5sFcTg
Via: SIP/2.0/UDP 10.x.x.x:15060;rport=15060;branch=z9hG4bK1433Za0Q89gHH
From: "2318232738*970742*112*9080" <sip:2318232738@10.x.x.x>;tag=F207Kct82BDgK
To: <sip:3151@ca1acd03.infonxx.local;sipx-noroute=VoiceMail;sipx-userforward=false>;tag=a111e48
Call-Id: c3a27f5f-2d83-122f-4398-005056ae006e
Cseq: 15250932 INVITE
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#56, [09:59:52.106] 45 Ms, To: 10.x.x.x:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.x.x.x:5060;received=10.x.x.x;branch=z9hG4bK-XX-13e44oV`61KkmxjGUEvk3KaZ2g
Via: SIP/2.0/UDP 10.x.x.x:5060;branch=z9hG4bK-XX-13e04qcW67ka_ASCscr`7znLrQ~zQUUUs9llK2c3ejN5sFcTg
Via: SIP/2.0/UDP 10.x.x.x:15060;rport=15060;branch=z9hG4bK1433Za0Q89gHH
Record-Route: <sip:10.x.x.x>
From: "2318232738*970742*112*9080" <sip:2318232738@10.x.x.x>;tag=F207Kct82BDgK
To: <sip:3151@ca1acd03.infonxx.local;sipx-noroute=VoiceMail;sipx-userforward=false>;tag=a111e48
Call-ID: c3a27f5f-2d83-122f-4398-005056ae006e
CSeq: 15250932 INVITE
Contact: <sip:3151@10.x.x.1:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 173
Content-Type: application/sdp
v=0
o=LanScape 3520162792 3520162792 IN IP4 10.x.x.1
s=LanScape
c=IN IP4 10.x.x.1
t=0 0
m=audio 3044 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#140, [09:59:52.152] 110 Ms, From: 10.x.x.x:5060) <<<<
ACK sip:3151@10.x.x.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.x.x.x;branch=z9hG4bK-XX-13e6_cRuZkYX4bqHMtcWYJXJzw
Via: SIP/2.0/UDP 10.x.x.x:15060;rport=15060;branch=z9hG4bK2DXv15gU5j73c
Max-Forwards: 20
From: "2318232738*970742*112*9080" <sip:2318232738@10.x.x.x>;tag=F207Kct82BDgK
To: <sip:3151@ca1acd03.infonxx.local;sipx-noroute=VoiceMail;sipx-userforward=false>;tag=a111e48
Call-Id: c3a27f5f-2d83-122f-4398-005056ae006e
Cseq: 15250932 ACK
Contact: <sip:mod_sofia@10.x.x.x:15060;x-sipX-nonat>
Content-Length: 0
Date: Wed, 20 Jul 2011 14:59:54 GMT
Route: <sip:10.x.x.x>
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Thanks,
Paul
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PaulR
Intermediate
Joined: June 24 2011
Location: Philippines
Posts: 9
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| Posted: July 21 2011 at 1:09pm | IP Logged
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By the way, our current issue right now is some call drops for about 30 seconds. We suspecting the record-route header of 200 OK of Lanscape is incomplete.
Please advise.
Thanks,
Paul
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: July 22 2011 at 8:08am | IP Logged
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Hi Paul,
Record-Route:
I see what you mean in the 200 OK that the LME is sending back to the FreeSwitch. Will have to look into this. It does appear the LME is stripping off the additional parameters. I will repost when I have more info.
SipModifySipMessage:
Modifying transmitted SIP messages should work. I will verify this too.
Call Drops:
Question 1:
Do you mean that calls are “somehow” terminated after 30 seconds once they are connected?
That would be strange seeing the switch sent back the final INVITE ACK to the LME’s 200 OK response. I will test here to see if we get the same issue.
Question 2:
Is there a logging capability on FreeSwitch that can be enabled that will give us more information as to why the call is terminating?
Question 3:
I assume FreeSwitch is terminating the call. Yes?
Question 4:
Can you send me the LME SIP log via email for a dropped call?
Thanks,
RJ
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PaulR
Intermediate
Joined: June 24 2011
Location: Philippines
Posts: 9
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| Posted: July 22 2011 at 9:13am | IP Logged
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Hi RJ,
Please see my answer below.
SipModifySipMessage:
Modifying transmitted SIP messages should work. I will verify this too.
[Paul] I just read this post
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=591&PN =1, Randal says:
If you do not want to use base64 encoding/decoding as part of your SIP encode/decode scheme, simply alter Call-ID: headers to some other name and you will get the results you need. For example, when you encode your SIP message request headers, also be sure to change the Call-ID: header in the same SIP message to something else like x-Encoded-Call-ID:. This will force the media engine to handle the processing of your encoded SIP message differently.
I tried altering the "Call-ID" to xCall-IDx: and it works, the modified Sip message was successfully sent to acd BUT Acd does not recognise it
maybe becuase of the call-id header name was altered. Based on that post also , there are 2 ways to successfully modified the sip messages
1. Alter Call-ID header
2. Base64 encode
Question:Is there anyway to modified the sip messages without altering "Call-ID" and do any encoding?
Question 1:
Do you mean that calls are “somehow” terminated after 30 seconds once they are connected?
[Paul] Yes
Question 2:
Is there a logging capability on FreeSwitch that can be enabled that will give us more information as to why the call is terminating?
[Paul]I'll check on this and will let you know
Question 3:
I assume FreeSwitch is terminating the call. Yes?
[Paul]Yes
Question 4:
Can you send me the LME SIP log via email for a dropped call?
[Paul]Okay sure,will send the log in awhile
Thanks,
Paul
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: July 22 2011 at 1:56pm | IP Logged
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Paul,
Record-Route header optional parameters being stripped:
The optional parameters on received Record-Route headers are indeed being stripped via the LME SIP parsing code. Internally the optional parameters are being parsed properly but when SIP messages are finally constructed (to ASCII), the optional parameters are not added back to the Record-Route header just before the SIP message gets transmitted. We will look into this further and fix. This will be a simple fix.
Note: The ModifySipMessage() API is not at fault here and you are probably doing everything correct.
Dropped Calls:
I need your SIP log and a simple explanation of your deployment you are testing. Do you have a session border controller (SBC) or some other SIP proxy in front of your FreeSwitch?
I am not sure the record route header issue above is causing the problem. If FreeSwitch is terminating the call, that usually means that either FreeSwitch did not receive the final ACK or The “Route:” header in the ACK is causing SBC/Proxy/FreeSwitch to mishandle the ACK. We need the Sip logs to determine this.
You could also upload a WireShark network capture of your failed call to your support FTP account if you wish.
Enabling Record-Route: header in FreeSwitch:
I am not an expert when is comes to configuring FreeSwitch. Is there a quick way to enable the use of Record-Route headers in FreeSwitch?
I tested the LME against our latest git FreeSwitch install here and your server from a few week ago (extensions 4108, 4109). None of them are generating Record-Route headers when they send a call to the LME clients. If you can quickly tell me what to enable on FreeSwitch to enable the use of Record-Route headers, that will speed up the process.
Thanks,
RJ
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: July 22 2011 at 4:13pm | IP Logged
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Paul,
We have updated the LME with the Record-Route header parameter fixes we need.
I would like to verify that everything works so we do not have to go through multiple iterations of the fix. Can I test it against one of your servers?
Oh yes – Unless I have missed something, FreeSwitch does not need or support Record-Route headers seeing its not a true proxy so forget my previous question regarding enabling Record-Route headers.
Thanks,
RJ
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: July 25 2011 at 10:26am | IP Logged
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Paul,
Thanks for the test sequence you emailed. I will try to verify today and post back with the results.
RJ
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: July 25 2011 at 12:54pm | IP Logged
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Paul,
I placed an updated LME image in your support FTP account. It may be quicker for you to try this image directly.
The 866 number I call does not generate a received INVITE at my end.
RJ
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: July 26 2011 at 8:03am | IP Logged
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From Paul via email:
-----------------------
Hi RJ,
Sorry about that! Okay I'll download that updated LME and test it, then I'll let you know the result.
Thanks,
Paul
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PaulR
Intermediate
Joined: June 24 2011
Location: Philippines
Posts: 9
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| Posted: July 26 2011 at 1:01pm | IP Logged
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Hi RJ,
Its perfect! I can see a good Record Route now. Many Thanks!
Code:
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#56, [01:57:44.488] 7810 Ms, From: x.x.x.14:5060) <<<<
INVITE sip:4107@x.x.x.24:5060;x-sipX-nonat SIP/2.0
Record-Route: <sip:x.x.x.14:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2Aauth%7E.%2Afrom%7ERmdLRk43Y2F0OTZjbQ%60%60.900_ntap%2Aid%7EMzIxOTgtMTc0Mg%60%60%21a9d50b0d30cb0f987671027908558654>
Via: SIP/2.0/UDP x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg;id=32198-1742
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
Max-Forwards: 18
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 INVITE
Contact: <sip:mod_sofia@x.x.x.14:15060;x-sipX-nonat>
User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110326T074140Z
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 201
X-Dnis: 17127
X-Fs-Support: update_display
Remote-Party-Id: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;party=calling;screen=yes;privacy=off
Date: Tue, 26 Jul 2011 17:49:51 GMT
Expires: 20
X-Sipx-Handled: X10.112.75.14-x.x.x.14
v=0
o=FreeSWITCH 1311689843 1311689844 IN IP4 x.x.x.14
s=FreeSWITCH
c=IN IP4 x.x.x.14
t=0 0
m=audio 12748 RTP/AVP 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#57, [01:57:44.597] 109 Ms, From: x.x.x.14:5060) <<<<
INVITE sip:4107@x.x.x.24:5060;x-sipX-nonat SIP/2.0
Record-Route: <sip:x.x.x.14:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2Aauth%7E.%2Afrom%7ERmdLRk43Y2F0OTZjbQ%60%60.900_ntap%2Aid%7EMzIxOTgtMTc0Mg%60%60%21a9d50b0d30cb0f987671027908558654>
Via: SIP/2.0/UDP x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg;id=32198-1742
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
Max-Forwards: 18
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 INVITE
Contact: <sip:mod_sofia@x.x.x.14:15060;x-sipX-nonat>
User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110326T074140Z
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 201
X-Dnis: 17127
X-Fs-Support: update_display
Remote-Party-Id: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;party=calling;screen=yes;privacy=off
Date: Tue, 26 Jul 2011 17:49:51 GMT
Expires: 20
X-Sipx-Handled: X10.112.75.14-x.x.x.14
v=0
o=FreeSWITCH 1311689843 1311689844 IN IP4 x.x.x.14
s=FreeSWITCH
c=IN IP4 x.x.x.14
t=0 0
m=audio 12748 RTP/AVP 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#43, [01:57:44.707] 8029 Ms, To: x.x.x.14:5060) >>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.14:5060;received=x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14:5060;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 INVITE
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.19 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#44, [01:57:44.707] 0 Ms, To: x.x.x.14:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.14:5060;received=x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14:5060;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>;tag=20aac7
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 INVITE
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.19 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#45, [01:57:44.707] 0 Ms, To: x.x.x.14:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.14:5060;received=x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14:5060;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>;tag=20862e
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 INVITE
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.19 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#58, [01:57:44.800] 203 Ms, From: x.x.x.14:5060) <<<<
INVITE sip:4107@x.x.x.24:5060;x-sipX-nonat SIP/2.0
Record-Route: <sip:x.x.x.14:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2Aauth%7E.%2Afrom%7ERmdLRk43Y2F0OTZjbQ%60%60.900_ntap%2Aid%7EMzIxOTgtMTc0Mg%60%60%21a9d50b0d30cb0f987671027908558654>
Via: SIP/2.0/UDP x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg;id=32198-1742
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
Max-Forwards: 18
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 INVITE
Contact: <sip:mod_sofia@x.x.x.14:15060;x-sipX-nonat>
User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110326T074140Z
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 201
X-Dnis: 17127
X-Fs-Support: update_display
Remote-Party-Id: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;party=calling;screen=yes;privacy=off
Date: Tue, 26 Jul 2011 17:49:51 GMT
Expires: 20
X-Sipx-Handled: X10.112.75.14-x.x.x.14
v=0
o=FreeSWITCH 1311689843 1311689844 IN IP4 x.x.x.14
s=FreeSWITCH
c=IN IP4 x.x.x.14
t=0 0
m=audio 12748 RTP/AVP 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#46, [01:57:44.800] 93 Ms, To: x.x.x.14:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.14:5060;received=x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14:5060;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>;tag=2087c9
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 INVITE
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.19 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#59, [01:57:45.488] 688 Ms, From: x.x.x.14:5060) <<<<
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#47, [01:57:45.800] 1000 Ms, To: x.x.x.14:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.14:5060;received=x.x.x.14;branch=z9hG4bK-XX-92a8ghGuNI1hhdADF1B2LgIhXA
Via: SIP/2.0/UDP x.x.x.14:5060;branch=z9hG4bK-XX-92a4msP0kH4kgYY8vDfWe9tokw~iw_VVV8l9d2bqU8d9t5CXg
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKXpFm5c9r13FrQ
Record-Route: <sip:x.x.x.14:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2Aauth%7E.%2Afrom%7ERmdLRk43Y2F0OTZjbQ%60%60.900_ntap%2Aid%7EMzIxOTgtMTc0Mg%60%60%21a9d50b0d30cb0f987671027908558654>
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>;tag=2087c9
Call-ID: 80d2fc72-3252-122f-1c85-005056ae004f
CSeq: 15515231 INVITE
Contact: <sip:4107@x.x.x.24:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
User-Agent: IDEA Application using Lanscape Media/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.19 (www.LanScapeCorp.com)
Content-Length: 231
Content-Type: application/sdp
v=0
o=LanScape 3520691865 3520691865 IN IP4 x.x.x.24
s=LanScape
c=IN IP4 x.x.x.24
t=0 0
m=audio 3982 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 (#60, [01:57:45.956] 468 Ms, From: x.x.x.14:5060) <<<<
ACK sip:4107@x.x.x.24:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.14;branch=z9hG4bK-XX-92aa8ur0x7koCt6G9QjVl9M7zw
Via: SIP/2.0/UDP x.x.x.14:15060;rport=15060;branch=z9hG4bKyZ8c77Svyc6aK
Max-Forwards: 20
From: "6108491500*568355*113*17127" <sip:6108491500@x.x.x.14>;tag=FgKFN7cat96cm
To: <sip:4107@ca1acd04.infonxx.local>;tag=2087c9
Call-Id: 80d2fc72-3252-122f-1c85-005056ae004f
Cseq: 15515231 ACK
Contact: <sip:mod_sofia@x.x.x.14:15060;x-sipX-nonat>
Content-Length: 0
Date: Tue, 26 Jul 2011 17:49:52 GMT
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Thanks,
Paul
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Record-Route question
