Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: alexandercoachman on September 27, 2011, 03:50:13 PM
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Hi all!
I have a problem, please help me.. I have a public Genesys SIP server with Stream Manager. I use the Requestec's Flash2SIP gateway, which is integrated to our Genesys environment. It converts the real-time SIP media to flash, so customers doesnt need any softphones, only flash support in their browser. They click on a link, a flash area opens, and they can see a video.
This works. (With not so good video quality)
So i created a routing strategy with two play announcement blocks, so i created a loop. When the first video ends, comes the another and so on...
But when the first video ends, the second video fails, we can hear only the voice of the video. I captured this issue with wireshark on the SIPServer side. When the second video starts, there are many [b]Destination unreachable (Port unreachable)[/b] ICMP messages with source of my SIPServer and destinated to the public Flash2SIP gateway.
Here is the link, where you can test my solution in flash:
[b]Sorry, the link is unavailable for you now because, we are testing today.[/b]
Here are the SIPServer, Stream Manager logs, and the capture file:
[url=http://www.sendspace.com/file/b434gw]http://www.sendspace.com/file/b434gw[/url]
Please, tell me your ideas..
Alex.
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I found this error in Stream Manager logs:
Probing announcement/11 (codecs=[b]u.....3[/b].,fmt=wav)
[b]found announcement/11.zip -> pcmu h263_CIF=2 Ok[/b]
Probing announcement/11 (codecs=[b].a39%..[/b].,fmt=wav)
[b]found announcement/11.zip -> media not found[/b]
???
First found, then immediately not found? ??? Whats the second difference in codecs? (u.....3 vs. .a39%..)
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I found the followings in the SIPServer logs when the second video starts:
17:33:33.343:(1) SIPS:LOGBLOCK:BEGIN:SIPDATA:[
17:33:33.343:(1) Received [424,TCP] 455 bytes from 10.35.37.10:1190 <<<<<
[b]BYE sip:v33245@10.35.37.10:5060;[/b]transport=tcp SIP/2.0
From: <sip:annc@10.35.37.10:5065;play=announcement/11;repeat=1>;tag=65701626-E872-40F5-8C13-C857465EF25E-1
[b]To: "Sandor" <sip:v33245@213.152.241.166>;[/b]tag=72BB95EC-30C4-44A5-A681-36810ED69348-2
Why appears my Requestec account with two different IP address? Is this normal? The internal IP is my SIPServer IP address, the public is the Flash2SIP gateway IP.
The following important events:
17:33:33.3430 [0] 8.0.400.25 [b]distribute_response: message EventTreatmentEnd[/b]
17:33:33.343 Int 04544 Interaction message [b]"EventTreatmentEnd" generated[/b]
17:33:33.343 Trc 04542 [b]EventTreatmentEnd sent to [380] (00000003 URS 10.35.37.10:1141)[/b]
17:33:33.343 Trc 04542 [b]EventTreatmentEnd sent to [436] (00000004 StatServer 10.35.37.10:1178)[/b]
Here comes back the URS, because the call arrived the second play announcement block:
17:33:33.343 Trc 04541 [b]RequestApplyTreatment received from [380] (00000003 URS 10.35.37.10:1141)[/b]
17:33:33.343 Int 04543 [b]Interaction message "RequestApplyTreatment" received from 380 ("URS")[/b]
Next thing what I dont understand, the local contact (it appears again with the SIPServers internal IP address)
17:33:33.343 SIPCONN(Treatment Service): Local contact: '<[b]sip:v33245@10.35.37.10:5060[/b];transport=tcp>'
Alex.
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When I make calls from a Softphone, it plays the video correctly. This softphone is installed on the SIPServer and registered as an Extension. :-\
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We are using an ASA5505, configured with static NAT. I think the problem will be located on this firewall.
What do you think about it?
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SIP is not NAT friendly...you should use some SBC with that
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Agreed. Have a look at AudioCodes Mediant. I have been looking into these in terms of SRTP with GVP 7.6 ....
http://genesysguru.com/blog/blog/2011/09/14/implementing-secure-voice-using-secure-rtp-srtp/
http://genesysguru.com/blog/blog/2010/11/19/avaya-call-classification/