SIP Call with Tandberg C40

[SchurmanRyan]

I have a Cisco Tandberg C40 codec and have been placing H.323 calls to an organization without any problems.  The organization is now making changes and requiring that I connect to them with their new SIP address.  When I try to connect, it immediately says the call could not be connected.  What are the requirements to place a SIP call with the Tandberg C40?

[dpetrovi]

Hi Ryan,

This support community is observed by MeetingPlace and Cisco WebEx Meetings Server specialists. I would advise you to post your question about Tandberg C40 into TelePresence support community as that community is observed by specialists for C40 and other Tandberg end-points:

https://supportforums.cisco.com/community/netpro/collaboration-voice-video/telepresence

I hope this will help you.

Thank you.

-Dejan

[Jens Didriksen · Accepted solution]

Assuming your C40 has software version TC6.x;

go to Network Services found under System Settings and turn on SIP.

Remember to make sure SIP is selected as the call protocol when calling this company though, default call protocol can only be set to either H.323, SIP or H.320, so if you only make SIP calls then set this to SIP, but if you predominantly make H.323 calls, then set this to H.323 and select SIP from the call options at the time of dial - or, even easier, add this address to the local phonebook with call protocol specified as SIP.

Just turning on SIP will allow you to make outbound SIP calls, however, if this company wants to call you, then they can only call you by using the public IP address of your system - and they might not even be able to do that depending on their deployment, policies etc.

If you want people to be able to call you using a SIP URI like myoffice@mycompany.com, then you need to register your system with a SIP registrar, i.e. CUCM, VCS-C etc, and also have SIP SRV records in place.

See the admin guide for more information:

http://www.cisco.com/en/US/docs/telepresence/endpoint/codec-c-series/tc4/administration_guide/profile-c60_codec-c60-c40_administrator_guide_tc40.pdf

/jens

Please rate replies and mark question(s) as "answered" if applicable.

[Martin Koch]

In addition to what Jens said. (and I always agree if somebody posts a link to the admin guide :-)

But it might be a bit much to comprehend if you are just a user and you have only a single system.

If you need to stick with an independent system.

As SIP on a stand alone device is not really NAT friendly I would recommend to check if its possible

to place it on a firewalled public IP without any NAT.

There are some Firewalls/NatHelper which can do a good job as well, but that causes often more trouble.

What is a nice thing is to have your system connected to a VCS. That is doing a lot of magic when you place a call.

For one endpoint its a bit much, but there are also companies who offer a registration as a service.

If you do a lot of video conferencing and you want to do more this might be worth investigating.

Please remember to rate helpful responses and identify helpful or correct answers.

[SchurmanRyan]

I do have a standalone box as Martin referred to above.  I have SIP enabled and have entered an external DNS server but still can't call a SIP address with a URI.  Any other ideas?

[Jens Didriksen]

Try a known "good" test site to confirm you can actually connect to anything at all using SIP;

loopback@rtp.ciscotac.net is such a site.

It wlll loop your video back to you if are successfull.

If you can connect to that site then chances are the problem is with that company you've been trying to connect to.

If you can't connect to that test site then you've got a problem your end, can you confirm your system has a public IP address and is not NAT'ed, and what software version is your C40 running?

Are you the only one who can't connect using the new SIP address, or are they having this problem with other external sites as well?

Does the domain in the SIP URI resolve properply? Do a nslookup and if it does come back with an IP address, try replacing the domain suffix in the SIP URI with that IP address. I.e. if it is myoffice@mycompany.com and mycompany.com resolves to 123.123.123.123 then try calling using myoffice@123.123.123.123

Do they have proper SIP SRV records in place? You can also check that by using nslookup.

/jens

Please rate replies and mark question(s) as "answered" if applicable.

[SchurmanRyan · Accepted solution]

Turns out I had to turn "TLS Verify" to "Off" and my call went through.  Using another test URI I was able to finally pinpoint this.  Thanks to everyone for the troubleshooting help.

[Martin Koch · Accepted solution]

Yes, that is the default setting, do you know why you enabled it?

Maybe one standard suggestion could also be, start from scratch with a factory default ;-)

And Ryan, please remember to rate helpful responses and identify helpful or correct answers!

[SchurmanRyan]

I'm not sure why or how it got enabled, but you are correct that verifying the default settings would be a good place to start.  Thanks again!

[Jens Didriksen]

Thanks for sharing the solution - +5 for that

/jens

Please rate replies and mark question(s) as "answered" if applicable.

[Martin Koch]

To minimize possible trouble makers is always the goal in trouble shooting.

That can include here:

* test with a "known to work site" (like Jens mentioned)

* check the remote address (also like Jens said)

* check with the remote site

* recheck config

* put the device on a public ip without nat

* try a different firewall

* disable all NAT helper / l3 features

* let somebody has a similar setup try to dial the number

* ...

If you do not know how to check the remote address, you can send it to me in a private message,

I could look it up (might be a good idea to not post it in the forum :-)

Please remember to rate helpful responses and identify helpful or correct answers.

[Joseph Chirayath]

I have TANDBERG Codec C60 which is having issues connecting to WebEx CMR. The codec is running TC5.1.3.292001 and is registered on VCSC version X8.7.2. The call goes through and the audio part connects fine when connected to the CMR without video. I tried changing the SIP profile type to Cisco/Standard but it did not help. Internal calls between codecs registered on VCSC works fine. I am just having issues with SIP calls made externally and the call connects without video. Is this is something related to the firmware version which is running on the codec which is causing this issue.

I am able to use the Cisco Codec C20 which is running version TC6.2.1.69d401c on the same VCSC works fine when connected to the same meeting.

Looking forward for your valuable feedback.

Thank you.

[Gerald Knight]

I am not sure but the code version is pretty old. You may want to search on this forum, Cisco will ususally provide you newer code to fix known security issues in 5.x and even later versions. Maybe that will correct your issue anyway.

[Joseph Chirayath]

I am not sure if the current version that I am running on the codec is causing the issue. Is there a way that this can be confirmed. Currently I do not have the options keys available to go for an upgrade on the Tandberg code's
Thank you for your feedback.
Joseph.

[Gerald Knight]

I am that familiar with the trace you jeed to tell what is going on, but others on this forum will be.

If you open a ticket based on PSIRT security issues on that (and later releases) I think Cisco will provide you the release keys and access to the newer code. That is up to Cisco.