Ask the Expert: Cisco TelePresence for the Enterprise

[ciscomoderator]

Welcome to the Cisco® Support Community Ask the Expert conversation.  This is an opportunity to learn and ask questions about Cisco Telepresence® for the enterprise. 

Cisco experts Jaret, Fernando, and Fred will be covering all Cisco TelePresence products.  Topics include Cisco TelePresence endpoints and TelePresence infrastructure such as the Cisco TelePresence Video Communication Server (VCS), Cisco Expressway Series, Cisco Unified Communication Manager (CallManager), Cisco TelePresence Servers (MSE 8710, on Virtual Machine, etc.), MCU (MSE 8510, etc.), Cisco TelePresence Management Suite (TMS), and all other Cisco TelePresence related devices.

Jaret Osborne is an 8-year Cisco Advanced Services veteran.  In his Advanced Services tour, Jaret has covered all aspects of Cisco Unified Communications and TelePresence products, including both enterprise and service provider verticals. Most recently Jaret has been working with global service providers supporting their Cisco TelePresence as a Service offerings while also incubating new cloud services at Cisco.

Fernando Rivas is a Cisco Advanced Services NCE, starting in the Cisco Technical Assistance Center (TAC), 2007, on the Collaboration Technology Team mastering the Cisco Unified Communication  technologies and specialized in call control CUCM,VCS) and  conferencing (MeetingPlace, Telepresence). In 2011, he joined Cisco Advanced Services as a member of the Cisco Collaboration team and participated in several Cisco TelePresence and video-related technologies deployments. Currently he is a member of the Video Cloud Technology Team, supporting video exchanges in several and architecting new private video cloud solutions for large enterprises. Fernando holds a routing and switching CCIE® certification (22975).

Fred Mollenkopf  is a Cisco Advanced Services Network consulting engineer working at Cisco for the last 7 years. Fred has led some of the largest Cisco Unified Communication and Collaboration deployments done for Cisco customers and partners. Over 15 years’ experience in data networking with a specialization in Cisco Unified Communications in 2004. Currently he is a member of the SP Video Advanced Services Team, supporting SP video exchanges and the Cisco Telepresence solutions.  Fred maintains an active CCIE® in Voice (17521).

Remember to use the rating system to let Jaret, Fernando, and Fred know if you have received an adequate response. 

Because of the volume expected during this event, Jaret, Fred, and Fernando might not be able to answer every question. Remember that you can continue the conversation in the Collaboration, Voice and Video Community, under the sub-community TelePresence, shortly after the event. This event lasts through August 15, 2014. Visit this forum often to view responses to your questions and the questions of other Cisco Support Community members.

[jaosborn]

Hi Patrick, I've included a sample below.  Also note that x8 introduced the ability to support regular expressions...  Note that each line begins with < and ends with >.  For example, the line that starts with "<cpl xmls" wrapped to a total of 3 lines on this forum editor.  The actual end of that line is "cpl.xsd">"

Pre x8 CPL sample:

<?xml version="1.0" encoding="UTF-8" ?>
<cpl xmlns="urn:ietf:params:xml:ns:cpl" xmlns:taa="http://www.tandberg.net/cpl-extensions" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:cpl cpl.xsd">
<taa:routed>
<address-switch field="origin">
<address is="old-source@domain.com">
<taa:location clear="yes" url="new-destination@domain.com" source-url-for-message="new-source@domain.com">
</taa:location>
</address>
</address-switch>
</taa:routed>
</cpl>

Post x8 CPL sample:

<cpl xsi:schemaLocation="urn:ietf:params:xml:ns:cpl cpl.xsd">
<taa:routed>
<taa:rule-switch>
<taa:rule unauthenticated-origin=".*" destination="(.+)@cisco\.com>
<taa:location clear="yes" regex="(?!.*@ciscotp\.com)" replace="\1" source-url-for-message-regex="(.*)@(ciscotp)\.com" source-url-for-message-replace="\1.@\2.net">
<taa:proxy/>
</taa:location>
</taa:rule>
</taa:rule-switch>
</taa:routed>
</cpl>

[Patrick Sparkman]

I'm familiar with the VCS CPL, just doesn't help there were no examples for this particular rule.  Took me a few minutes to figure out how I needed to edit your example, but it works just like I wanted.

Thanks!

[gcaudle66]

I know this forum is old, but I am attempting to use your script to rewrite source SIP URIs. However it is not working for me. i am concerned the issue I am having is that the source URI is an IP Address instead of a domain name. Someone mentioned previously that the source must be a domain for it to parse correctly. Is this correct? below is a snip of my config....

<taa:rule unauthenticated-origin=".*" destination=".*">
<taa:location clear="yes" regex="(.*)" replace="\1" source-url-for-message-regex="(.*)@172.26.4.4" source-url-for-message-replace="\1@test.com">
<taa:proxy/>
</taa:location>
</taa:rule>

[Patrick Sparkman]

Is this call for inbound or outbound dialing?
Is the source authenticated or unauthenticated (check VCS search history)?

[gcaudle66]

Outbound, and no. 

• • Authenticated: False

Just to test, I set the devices SIP URI manually to use a domain name and it still doesnt manipulate the URI. Now my config looks like.....

<taa:rule unauthenticated-origin=".*" destination=".*">
<taa:location clear="yes" regex="(.*)" replace="\1" source-url-for-message-regex="(.*)@test.int" source-url-for-message-replace="\1@test.com">
<taa:proxy/>
</taa:location>
</taa:rule>

I am running VCS 8.7

[Patrick Sparkman]

Did some testing, appears the source address doesn't support replacing IPs, only domains.

[mailgunj]

Hey guys!

I have a few queries about Video calling.

1.Does a one to one video call use a conference bridge?

2. Is it possible to have a video call between 2 standalone SX-20s that use NAT to translate to dedicated public IP addresses without a bridge? I had come across an implementation where they had NAT the SX-20 IP to a public IP and by dialing the IP address of the Blue Jeans website, they could initiate a call using the Blue Jeans bridge.

Thanks in advance.

Regards,

Gunjan Jain

[jaosborn]

Hi Gunjan -

1. Not required.  If the endpoints can reach each other via IP, and they speak compatible protocols the endpoints can call each other directly.  This of course requires your call control platform (CUCM or VCS, or whatever you are using) to be properly configured to allow the endpoints to call each other directly.

2.Yes, this is possible.  However, it will require the use of ICE.  ICE is a part of a protocol set that allows communications in these types of topology using STUN and/or TURN.  There are several "gotchas" to making this work - namely all devices in the path (endpoint, call control, firewalls) must support STUN and/or TURN.  With TURN there is a deicated device that the endpoints will send their media to and then that device will combine the streams.

Hope that helps.

-Jaret

[mailgunj]

Hi Jaret,

Thanks a lot! The information you provided was extremely helpful.

Regards,

Gunjan

[wisit_jan]

Hi,

TX9000 call to Telepresence Server have TIP issue (not attempting TIP/MUX negotiation (insufficient tokens specified))

Network Diagram

TX9000 --registered--> CUCM10 --sip trunk--> Telepresence Conductor ----> Telepresence Server VM

CUCM: 192.168.1.11

Telepresence Conductor: 192.168.1.17

TX9000 can call to conference number which match Conference aliases on Telepresence Conductor. But TX9000 video call active only 1 screen. Telepresence Server show log

282     17:09:48.943      APP     Info     conference "2011" created
283     17:09:49.072      SIP     Info     Incoming call from 192.168.1.17:53110
284     17:09:49.073      APP     Info     call 24: new incoming SIP call from "1001@192.168.1.11"
285     17:09:49.155      APP     Info     call 24: "1001" now joined conference "2011"
286     17:09:49.172      TIP     Warning     call 24: not attempting TIP/MUX negotiation (insufficient tokens specified)
287     17:09:49.172      SIP     Warning     call 24: local failure - attempting to process capabilities while extended video channel is opening

What is tokens in Telepresence Server?

Please suggest me for resolve this issue. Thank you

[Fernando Rivas Martin]

Hello Jan

Can you share the version of Conductor you are using?

One thing you can quickly check is the following:

- The Conference Alias you are accessing must be associated with a Conference Template.
Can you verify if that Conference Template has the setting "Allow multiscreen" set to Yes?
If not set it to Yes.

- Tokens is what the TPS use to allocate port and licenses (or portion of licenses) for each participant of the call. The more tokens you request, the more licenses you consume when accessing the TPS.

Regards

Fernando

[wisit_jan]

Hi Fernando,

Conductor version is 2.3

Allow mutiscreen is Yes.

TPS have 8 Screen license

I just call from one TX9000 to Conference Alias Name. I think TX9000 consume 4 screen license that sufficient.

Are TPS need more license? or TX9000 need some certificate?

[jaosborn]

Hello -

Can you verify that your CUCM Region configuraion is configured properly to allow all three screens?

Also, double check your SIP Profile that is assigned to the SIP Trunk.

-Jaret

[wisit_jan]

็Hi Fernando and Jaret,

Thank you all for support

Now it's worked. Participant quality in Conference templates must be Full HD.(default is HD)