QoS for Cisco TelePresence 3000
What are the QoS requirments for the Cisco TelePresence 3000 connected to a Cisco 6500 running CAToS?
Thanks in Advance.
This doc should help you:
http://www.cisco.com/en/US/products/hw/switches/ps708/products_configuration_guide_chapter09186a00800c65c9.html
Similar Messages
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Cisco 6807 and 6800ia Swtich QOS for Cisco ip phones
Does anyone have an example of configuring a 6800ia switch port connected to a 6807VSS parent for cisco ip phones qos. Normally we'd use auto qos voip but auto qos is not supported on 6800IA switches.
Cant find any cisco documentation of what the IA switches port config should look like for a cisco ip phone.
Any help would be appreciated.
Thanks,
DaveI'd leave QoS alone.
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Configuring TACACs Server for Cisco VPN 3000
Does anyone know how to get to the configuration setting to specify a TACACs server?
You need to be very careful when setting
up this thing. If the AAA server is down
for whatever reason, you will NOT be able to
log into the Concentrator again. As far
as the VPn3k console is concerns, it will
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even though the AAA is up and running. In
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Ask the Expert: Cisco TelePresence for the Enterprise
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.Tenaro,
Additionally here are the most common login issues. Unfortunately this includes items related to Presence implementation but I commented where we did not use these in our lab setup for CUCM Phone Capabilities only.
Login Issues
Problem:
Jabber Unable to Sign-in Through MRA
Solution
This can be caused by a number of things, a few of which are outlined below.
1. Collaboration Edge SRV record not created and/or port 8443 unreachable
For a jabber client to be able to login successfully using MRA, a specific collaboration edge SRV record must be created and accessible externally. When a jabber client is initially started it will make server DNS SRV queries:
_cisco-uds : this SRV record is used to determine if a CUCM server is available.
_cuplogin : this SRV record is used to determine if an IM&P server is available.
_collab-edge : this SRV record is used to determine if MRA is available.
If the jabber client is started and does not receive an SRV answer for _cisco-uds and _cuplogin, and does receive an answer for _collab-edge then it will use this answer to try to contact the Expressway-E listed in the SRV answer.
The _collab-edge SRV record should point to the FQDN of the Expressway-E using port 8443. If the _collab-edge SRV is not created, or is not externally available, or if it is available, but port 8443 is not reachable, then the jabber client will fail to login.
2. Unacceptable or No Available Certificate on VCS Expressway
After the jabber client has received an answer for _collab-edge, it will then contact the expressway using TLS over port 8443 to try to retrieve the certificate from the expressway to setup TLS for communication between the jabber client and the expressway.
If the Expressway does not have a valid signed certificate that contains either the FQDN or domain of the Expressway, then this will fail and the jabber client will fail to login.
If this is occurring, the you should use the CSR tool on the Expressway, which will automatically include the FQDN of the expressway as a Subject Alternative Name.
MRA requires secure communication between the Expressway-C and Expressway-E, and between the Expressway-E and external endpoints.
Expressway-C Server Certificate Requirements:
The Chat Node Aliases configured on the IM&P servers. This is required if you are doing XMPP federation. The Expressway-C should automatically include these in the CSR provided that an IM&P server has already been discovered on the Expressway-C.
The names in FQDN format of all Phone Security Profiles in CUCM configured for TLS and used on devices configured for MRA. This allows for secure communication between the CUCM and Expressway-C for the devices using those Phone Security Profiles.
Expressway-E Server Certificate Requirements:
All domains configured for Unified Communications. This includes the domain of the Expressway-E and C, e-mail address domain configured for Jabber, and any presence domains.
The Chat Node Aliases configured on the IM&P servers. This is required if you are doing XMPP federation.
The MRA Deployment guide describes this in greater detail on pages 17-18. (http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-1/Mobile-Remote-Ac...
Note: In our lab for testing Phone Capabilities only, we did not include the Chat Node Aliases in the certificate as we were not using IM&P.
3. No UDS Servers Found in Edge Config
After the Jabber client successfully establishes a secure connection with the Expressway-E, it will ask for its edge config. This edge config will contain the SRV records for _cuplogin and _cisco-uds. If these SRV records are not returned in the edge config, then the jabber client will not be able to proceed with trying to login.
To fix this, make sure that _cisco-uds and _cuplogin SRV records are created internally and resolvable by the Expressway-C
More information on the DNS SRV records can be found on page 10 of the MRA deployment guide for X8.1.1 (http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-1/Mobile-Remote-Access-via-Expressway-Deployment-Guide-X8-1-1.pdf)
Note: In our lab for testing Phone Capabilities only, we did not include the DNS SRV for _cuplogin.
4. The Expressway-C logs will indicate the following error: XCP_JABBERD Detail="Unable to connect to host '%IP%', port 7400:(111) Connection refused"
If Expressway-E NIC is incorrectly configured, this can cause the XCP server to not be updated. If the Expressway-E meets the following criteria, then you will likely have this issue:
Using a single NIC
Advanced Networking Option Key is installed
Use Dual Network Interfaces option is set to “Yes”
To correct this problem, change the “Use Dual Network Interfaces” option to “No”
The reason this is a problem is because the Expressway-E will be listening for the XCP session on the wrong network interface, which will cause the connection to fail/timeout. The Expressway-E listens on TCP port 7400 for the XCP session. You can verify this by using the netstat command from the VCS as root.
Note: We used a Dual Network Interface Expressway for testing but were not using XCP, so this was not applicable to us.
5. VCE-E Server hostname/domain name does not match what is configured in the _collab-edge SRV.
If the Expressway-E Server hostname/domain name does not match what was received in the _collab-edge SRV answer, the jabber client will not be able to communicate to the Expressway-E. The Jabber client uses the xmppEdgeServer/Address element in the get_edge_config response to establish the XMPP connection to the Expressway-E.
This is an example of what the xmppEdgeServer/Address would look like in the get_edge_config response from the Expressway-E to the Jabber client:
<xmppEdgeServer>
<server>
<address>ott-vcse1.vcx.cisco.com</address>
<tlsPort>5222</tlsPort>
</server>
</xmppEdgeServer>
To avoid this, make sure that the _collab-edge SRV record matches the Expressway-E hostname/domain name. Enhancement CSCuo83458 has been filed for this.
Note: This was one of our issues when we first setup. We adjusted our Expressway-E to insure the below:
System > Administration > System Name this was the FQDN
System > DNS > System Host Name was the host portion of the FQDN
System > DNS > Domain Name was the domain portion of the FQDN
System > Clustering > Cluster Name (FQDN for Provisioning) was the FQDN
6. Unable to log into certain IM&P servers. VCS logs say "No realm found for host cups-example.domain.com, check connect auth configuration"
From the Expressway-E, go to Configuration -> Unified Communications -> IM&P Servers. Open each server and click "Save" again. Not sure exactly why this happens.
Note: This was N/A to our test and can be ignored with Phone Capabilities only.
Thanks
Fred -
Hi All,
Our problem is, we have Cisco Works LMS 3.0.1. cannot archieve configuration for cisco 3000 series vpn concentrator.
Any help would be greatly appreciated.
Thanks in advance.
SamirMake sure you have filled out all of the HTTP/HTTPS credential data in DCR for these devices. RME will only use HTTPS to fetch VPN concentrator configurations.
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VoIP QoS for Site-to-Site Tunnel
Hi all,
I need some help with setting up QoS for VoIP between two Cisco ASA 5505 with Site-to-Site VPN.
There is no need for bandwith reservation, only DSCP 46 (EF) should be highest and DSCP 26 second highest queue and the rules should only apply to a site-to-site VPN.
Usually i try to configure the ASAs via ASDM and found out in the Cisco Documentations how to setup QoS for the DSCP bits with a Service Policy and how to set up QoS for a Site-to-Site VPN (Service Policy Rule -> Traffic Match). But how do i configure the QoS for a DSCP bit to only apply to a Site-to-Site Tunnel? And how do i configure different priorities for the two DSCP bits, is this defined by the order of Service Policies?
The QoS has to be enabled on both ASAs for the inside interface?
Thanks in advance
TobiasHi Collin,
ok so from the document i think this has to be added on both ASAs to prioritize DSCP 46. Could you have a quick look if this is all i need?
class-map voice_traffic
match dscp ef
class-map data
match tunnel-group <Tunnel Name>
match flow ip destination-address
policy-map voice_policy
class voice_traffic
priority
class Data
police output 200000 37500
service-policy voice_policy interface outside
But there is still the question how two configure another DSCP bit for priority 2, I could not find out how this is done, neither CLI nor ASDM. Any ideas?
BR
Tobias -
Cisco 2960x - auto qos voip cisco-phone on access ports
After configuring "auto qos voip cisco-phone" on a stack of 2960x switches, we later noticed that after a power cycle (config was saved first) it went missing when doing a "show run" from the interface configs, on one of the switches (g1/0/1-48 had it applied, then after power cycle it didn't show). However, doing a "show auto qos interface" shows it is still applied to all interfaces g10/1-48.
I try to do "auto qos voip cisco-phone" again on g1/0/1-48, and it shows this:
SwitchStack01(config-if-range)#auto qos voip cisco-phone
AutoQoS Error: AutoQoS already configured
% Range command terminated because it failed on GigabitEthernet1/0/1
If I attempt to remove "auto qos voip cisco-phone", it shows this:
SwitchStack01(config-if-range)#no auto qos voip cisco-phone
AutoQoS Error: AutoQoS not configured
% Range command terminated because it failed on GigabitEthernet1/0/1
When I do a "show run", it is no longer displayed for the first switch of the stack.So is it still applied to the interfaces or not? Is this a bug? Has anyone else experienced this? This has happened to at least a couple stacks of 2960x's we have.
Thanks!It appears to be a bug in the code. Open a TAC ticket with Cisco and send them the output and the IOS version you are using so they can help you resolve the issue. Most likely, they recommend an upgrade.
HTH -
Cisco Telepresence and Meeting Place Integration with Third Party Call Pros
Dear colleages:
i want to ask if Cisco Telepresence can integrate with 3rd party call processing systems.
as my project has already existing Nortel Communication Server 1000,
which supports:
Operating System: VXWorks and Linux
Network Signaling Protocols: H.323v4, MCDN, SIP, LDAP, QSig, QSS, in band and DPNSS
Trunking Support:
IP:IP Virtual Trunks using IP Peer Networking over H.323 or SIP
Digital: DTI, ISDN-PRI, ISDN-BRI
Analog: Loop and Ground Start CO, FX, WATS, two or four wire E&M, four wire DX, DID, TIE, RAN Paging.
and i want to build Video Converence system over the WAN.
please to tell me if that integration is possible and what is the design components that is required for such project to be handled from A- to Z.
Regards,I have been told that i need to use SIP protocol API instead of TAPI to have a TRUE VOIP Windows solution.
Could you guys please tell me the difference between SIP API and TAPI?
And could you please suggest a way which API (SPI or TAPI) the OnCast solution is based on (http://www.litescape.com/oncastsoftware.html) - we have to develop something similar.
Thank you for your replies,
Alexey -
Passing args to: Cisco Telepresence Movi.msi
Hi -
I'm working with Movi 4.2 and am using the Cisco Telepresence Movi.msi because I'm creating my own installer.
I'm trying to pass these arguments to it for a silent install:
/quiet DOMAIN=transport-you.com EXTERNALVCS=transport-you.com HIDEADVANCEDLOGIN=1
The Movi_Administrator_Guide_4-2.pdf says I can set these properties but for some reason it's failing when I run it. Here's an example error msg:
The system cannot find the file DOMAIN=transport-you.com.
while executing
"::InstallJammer::actions::$component $this"
Error in action LaunchFile
What am I missing here?Hi Mattew!
Use the .exe installer file provided in the zip file!
The MSI would be used if you want to build your own software package.
From the admin guide: -
I have Cisco TelePresence SX10 and content sharing is not working when I am dialing through the bridge, I can share the content if I drag and drop from RMX, but if schedule the call in Resource manager or manually dial in from device the content is not going to other hand, I have tried to turn off encryption as well but still same issue. can you please help me out with this. I am from Lion co and purchased sx 10 recently.
regards
HemangCan you please provide us with a little more information on your systems and configuration / topology, such as, what call control are you using (Cisco VCS, CUCM, other?). what type of "Bridge" are you using (is this a Cisco MCU, or Cisco TelePresence Server, or other device?), what versions of the software are on each of the devices, etc. The more information we have about your environment will help us assist you better. But saying that, if all your core equipment isn't Cisco, you may have more luck in the forums for the manufacturer of such equipment (ie Polycom's Support Community).
Wayne
Please remember to rate responses and to mark your question as answered if appropriate. -
How to setup QoS for IP/Mac Address WRVS4400N
I need some help setting up QoS for two Cisco/Linksys devivces on my network.
Network devices are Comcast SB6120 modem into WRVS4400n (v2.0.2.1 firmware) that divides into one 24 port gigabit Netgear unmanaged switch, 1 port into AT&T Cisco Microcell, 1 port to HP Officejet 7700 wired printer. Unmanaged switch has approx 10 devices attached (including another 8 port unmanaged gigabit switch). 3 static leases are defined for the network for three devices (HP OJ 7700 printer, Cisco/Linksys PAP2 VOIP adapter, windows home server v1. DHCP is handled by WRVS4400n. Comcast service provides 12mbps of service upstream, 4mbps downstream.
Need to setup QoS for AT&T Microcell and PAP2 VOIP adapter. The real issue that I can't figure out how to configure is IP based priority access. I can do rate control for an IP address but can't figure out how to specify priority control for an IP.
Here are my the current config questions:
1. Bandwidth management page: For the microcell device plugged into port 1, do i set the default CoS to 4 or 1?
2. Bandwidth management page: Should the bandwidth management type be set to priority? I assume yes since I need to define the UDP/TCP ports for priority. The major issues here are
a) one of the ports the Microcell uses is secure https (TCP 443) which is could also be used by any other internet device.
b) can't specify priority for the PAP2, since it's behind the unmanaged 24 port switch
3. QoS setup page: Do I need just 2 queues (one for the port ID for the Microcell and one for the IP address of the Linksys adapter)?
4. QoS setup page: I assume I need to define DSCP for the unmanaged switch but not sure how to tie a queue to an IP address
4. QoS setup page: Shoud priority 0 be set for the most important queue (microcell) and the priority 1 CoS be for the VOIP adapter? Again, not sure about how to define priority by IP address.
5. DSCP setup page: As far as I can find, there's no documentation on how to setup this page.That's what I do. I keep it simple, one calendar, one set of contacts.
iCloud is really the simplest way to go, as it will sync your (eventual) single Contacts and Calendar across all your devices.
I would:
Figure out which single calendar (in Calendar.app) and contacts (in Contacts.app) that you want to keep.
I'd make a backup of each one. For example,
in Contacts, File > Export... > Contacts Archive...
in Calendar, File > Export... > Calendar Archive...
Make a backup of the other calendar events and contacts that you want to save.
delete all the contacts and calendars from your phones and your iPad.
add the contacts and events from the backups to the single calendar in Calendar.app and your contacts in Contacts.app on your iMac. There is probably some way to export the contacts and events you are adding, and then combine them, but you could also just add them manually.
Then when you configure all your devices for iCloud, they will be all synced automatically.
Message was edited by: arthur -
To configure a QoS for the attached topolgy
Dear friends,
I have to configure QoS for the attached topology with the following requirements, can any one please help me to do the same.
The minimum requirements are that 4 simultaneous VoIP calls need to be supported across the backbone, as well as a dedicated 10MBs of http traffic. Of the remainder of the bandwidth ftp traffic should be given a high priority. Make sure routing protocols are given a suitable priority level.
Thanks and expecting repliesHere are some questions that you'll need to answer, to get started.
What does the HTTP requirement mean? It is not possible to get 10 Megabytes per second on a 1.544 Megabit per second line.
What voice codec will be used on the IP Phones? This will determine how much bandwidth you need to allocate.
Next, read the following link:
http://www.cisco.com/en/US/docs/ios/qos/configuration/guide/12_4t/qos_12_4t_book.html
It's all in there. Unfortunately, it's a big topic, so there is a lot there. -
Hello,
I am very new to voip, second day. I have configured a 3850 stack to trust cisco phones on the switchports with the auto qos feature. I am interested to know where else I have to apply qos for the policy's to apply. The 3850 stack is uplinked to a 4500x "core". My question is once the qos is applied at the switch port, is everything just grand and my qos will work or does it have to be applied elsewhere.
Thank you.
DallasIt has to be applied all throughout your network. Ie Core switch, access layer, uplinks to voice servers/gateways and access ports to phones. If you have wireless phones, those will have to be considered as well.
-
3850 - auto qos voip cisco-phone
I am having an issue where it appears that the "auto qos voip cisco-phone" command is marking down my voice packets COS and DSCP values to 0. Anyone else notice this? I am connecting Cisco 7945 phone to the interfaces with this command applied to the interface. I verified that the IP Phone is sending COS = 5 and DSCP = ef to the switchport. I verified the packets inbound to the switchport and the packets outbound as they transit to the destination IP Phone's switchport.
Hi joshua,
What is the IOS-XE version you are running on your 3850 ?
Post your switchport configuration (7945 phone connected)
Here is 3.3 QoS config guide & I noticed it says something like below qouted. So make sure this trust device command is there & your port is configured for standard VoIP port connectivity.(like voice vlan, etc)
http://www.cisco.com/c/en/us/td/docs/switches/lan/catalyst3850/software/release/3se/consolidated_guide/configuration_guide/b_consolidated_3850_3se_cg/b_consolidated_3850_3se_cg_chapter_01101000.html#reference_8A1C35AB2874498F90C4F5600301BE9D
"The trust device device_type interface configuration command is only supported in an auto-QoS configuration, and not as a stand-alone command on the switch. When using the trust device device_type interface configuration command in an auto-QoS configuration, if the connected peer device is not a corresponding device (defined as a device matching your trust policy), both CoS and DSCP values are set to "0" and any input policy will not take effect"
HTH
Rasika
***** Pls rate all useful responses **** -
We have some LNSs running 12.4 SP Services, running with the following config (qos specific lines and currently not working):
class-map match-all voice-signaling
match access-group 101
class-map match-all voice-traffic
match access-group 102
policy-map sub-policy
class voice-traffic
priority 240
class voice-signaling
bandwidth 16
policy-map XXX_qos-voice
class class-default
shape average 256000
fair-queue
service-policy sub-policy
interface Virtual-Template1
ip unnumbered Loopback0
ip mroute-cache
no peer default ip address
ppp authentication chap
no clns route-cache
access-list 101 remark -- SCCP/H323/MGCP/SIP --
access-list 101 permit tcp any any range 2000 2002
access-list 101 permit tcp any any eq 1720
access-list 101 permit tcp any any range 11000 11999
access-list 101 permit udp any any eq 2427
access-list 101 permit udp any any eq 4569
access-list 101 permit udp any any eq 5036
access-list 101 permit udp any any eq 5060
access-list 102 remark -- RTP Traffic --
access-list 102 permit udp any any range 16384 32767
in the feature set for the IOS currently running it suggests this configuration should work, however it will not apply the policy via the Cisco-AVPair command (the command is visible via the debug AAA per user so radius is working, even when we test it by applying the service-policy to the actual virtual-template it errors saying it will only work on an MPL bundle. From what I have read it is suggested that even though this feature is supposed to work, it does not in practice and an upgrade to a feature set which includes QoS:per session shaping and queueing on LNS is needed.
If anyone has any experience of the or has any suggestion of if we can achieve our QoS for voice per session with the current IOS that would be great.This feature is not related to the NPE type. This feature is performance impacting so you should do some tests to see if your NPE-400 can support the number of L2TP tunnels you want with this feature enabled. If it's not the case, you will have to upgrade your NPE.
HTH
Laurent.
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