03-18-2012 06:58 AM - edited 03-17-2019 10:56 PM
Dears,
I have 2 Tandberg Units. C20 and EX90. I am trying to establish a call between them over the internet. I made the required NAT. and also configure the callsetup to direct and the NAT address on both units. when i establish a call, i get the ringing on the other side but when i accept the call is dropped.
Can you please help.
Best Regards,
Abbas
03-18-2012 03:03 PM
While I would need to know more about your specific configuration, it sounds like your call signalling works but the media does not.
I imagine both your endpoints are registered to VCS? Are they registered to the same VCS or seperate?
For example, if they are registered to different VCS then it sounds like the VCS's can communicate but one or both erndpoints can't route to each other (or are getting blocked or incorrectly NAT'd).
03-19-2012 06:04 AM
Hi Abbas
Do you have same settings for encryption (Advanced Configuration>Conference1>Encryption) for your both terminals?
Br. Oleksandr
03-19-2012 08:32 AM
Abbas/Nick:
I would assume a VCS is not used here, if a VCS-E is used no NAT setting on the endpoint shall be done,
if a VCS-C, not sure if static set up nat is properly supported.
Besides the encryption failure Oleksandr mentioned I would also check the firewalls, that you have
proper 1:1 nat, the needed ports are forwarded and that no ALG or other NAT helper kicks in.
Besides that analyzing log files and a network trace can be helpful.
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03-19-2012 08:57 AM
Hi Martin,
Yes, there is no VCS in the middle. thats direct call between them. after I edited some setting i can get audio only now and from call status i can see that audio negotiation is done and the audio codec is g711. but for the video codec, it is not negotiated.
Best regards,
Abbas
03-19-2012 09:52 AM
Are you trying to use SIP or h323?
For a direct ip2ip call I would use h323, you hace to set the call mode to direct for it,
I would disable sip, its not that common used in the vc world for ip2ip calls.
In general I can recomend using a VCS-E for firewall traversal, either your own or use a
service who can provide you with that.
I do not see a need to "edit some settings" regards the codecs if you just want to to a ip2ip
call in between these two endpoints.
Not sure what else you "edited", so in doubt start over with doing a factory reset if you lost
control why and what you changed.
For such a call and suitable bandwidth I would expect the call to use audio: aac-ld and video: h.264
Its not much configuration needed to bring the systems up:
* have ip connectivity
* enable 1:1 nat from the public ip address of the router to the IP of the endpoint
* enable h323 (xConfiguration NetworkServices H323 Mode: On)
* enable direct h323 mode: (xConfiguration H323 Profile 1 CallSetup Mode: direct)
* check that sip is disabled (NetworkServices sip Mode: off
* and h323 is the default protocol (xConfiguration Conference 1 DefaultCall Protocol: H323)
* define the external address (xConfiguration H323 NAT Address: "
* enable NAT mode ( xConfiguration H323 NAT Mode: on )
If you still experience strange symptoms check that the firewall does not have a nat helper,
ALG, ... whatever active.
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03-20-2012 12:11 AM
Hi All,
There is no need of VCS in this call setup , you just need to open Media Ports on your firewall so that call signalling and Media can flow through.
I would like to suggest please check the firewall settings at both the ends since there need to be some ports open on the firewall for this to work.
The port numbers that are needed to be opened on the firewall are as follows:
For H.323:
• Gatekeeper Discovery (RAS) – Port 1719 – UDP
• Q.931 call Setup – Port 1720 – TCP
• H.245 – Port Range 5555-5574 – TCP
• Video – Port Range 2326-2485 – UDP
• Audio – Port Range 2326-2485 – UDP
• Data/FECC – Port Range – 2326-2485 – UDP
For SIP:
SIP messages – Port 5060 – UDP/TCP
• SIP messages – Port 5061 – TLS(TCP)
• H.245 – Port Range 5555-5574 – TCP
• Video – Port Range 2326-2485 – UDP
• Audio – Port Range 2326-2485 – UDP
Regards,
Saurabh
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