VoIP Deployment

My company is going to deploy Voip to the network. We have about 10 major sites with about 10,000 phones. I hope someone could give me suggestion of what tools I should use for pre-deployment; for exampele, NetIQ or NetQos to analysis and remediate the network before we put ip telephony to the network.
Any recommendation would be appreciated.
Thanks

I think the network piece of the picture is the easy part, its the documention of the existing PBX system configuration
- ACD groups
- number of lines on a phone. (determines 7940 or 7960 series to purchase, or even 7914 options)
- voicemail migrations to Unity platform.
- testing a dual platform with CCM running and old PBX.
- PSTN services.. what stays, what goes
- analog support.... modems, faxes, 1mbs, etc.
- what type of training to support or provide
- headset replacements
- tie lines and existing routing patterns to other sites
I have been at a client for the past 6 months preparing just preparing for migrating 1000 phones... at one site.... from a 15 year PBX.... it's much bigger than anyone every thinks.
drop me a line if you need some advice
cheers!

Similar Messages

Voip deployment over mpls link

Hi
we want to deploy voip for multiple location over the mpls lines we have between the multiple location can any body suggest the feasibility and how to do it and any related document and guidance for that
Thanks
Manish gaur

Manish,
That is probably the best way to connect multiple sites without having to connect each one with a T1, etc. Our MPLS vendor has an SLA on latency with 50ms or less guaranteed. With voice you want to make sure the latency is definitely less than 150ms. For ease of management we're using a router at each remote site with SRST capabilities so that all of the configuration remains at the main HQ callmanager, but if the connection ever goes down, the remote site phones will still be able to call each other (not main office until connection is restored) Alternatively you can put a router at each site with CallManager Express, you will just have to manage each site separately. Hope that helps a little.

Secured Voip Deployment

Hi,
Wanted to know how to deploy Cisco IP Phones over Secured network through Firewall?

without knowing too much about the secured network, the obvious answer would be to punch some holes in the firewall for SCCP/SIP and RTP
Cheers

VoIP Bandwidth

I was reading some articles online about multi-site VoIP deployment. Inside articles, they are using 64Kbps or less as bandwidth values between routers to show QoS, RSVP and CAC for Voice. This makes me think is it still relevant?
I mean we are in 2013 almost 2014 and I have Fibre to my house. Most of my customers are using or upgrading to 10Mbps (at least) ISP links. I personally never deployed a production Frame-Relay setup but what do I know...
Do you guys still have "dial-up" speed links running in production?

Inside articles, they are using 64Kbps or less as bandwidth values between routers to show QoS, RSVP and CAC for Voice. This makes me think is it still relevant?
Depends on the codec being used.

QoS Configuration for NME-AIR-WLC

Hello,
I have a question on the correct QoS configuration for a VoIP deployment. I have followed the steps referenced here but have two questions..
1) The last recommendation in the list is to set DCA channel sensitivity to High to avoid changes during business hours. I think this should be Low? The manual states "Low—The DCA algorithm is not particularly sensitive to environmental changes."
2) The recommendation is to trust dscp on AP uplinks and to trust cos on the WLC uplink. However I can't find any configuration guidelines for configuring QoS if the WLC in question is an NME hosted in a 2921.
2a) The link between the switch and the 2921 is an 802.1q trunk so should I trust cos on this link rather than dscp?
2b) What configuration should be on the 2921? Create a policy-map and apply it to the uplink port?
2c) Should there be a service policy assigned to the In1/0 interface for the WLC itself?
Any thoughts appreciated.
Thanks,
Dave

Well I appear to have answered question 2 after I stopped being lazy and did some packet captures.
No configuration required on 2921 as packets captured from In1/0 and Gi0/0 show correct DSCP markings on RTP and SCCP packets.
Would still be interested to know on question 1 though.

WAAS Express & CoS on same router

I have a location that has been running WAAS Express for a while. I have also implemented CoS for a pending VoIP deployment. Is there anything I need to do in the Manager to make these two work together? I have the Policy Manager and other items needed for the CoS to operate and I am seeing hits on my ACLs for remaring the traffic but my Solarwinds monitoring system is not picking up any differences from a Class of Service standpoint. Any ideas where I should start to troubleshoot this issue? How can I verify the re-classifications on the router / netflow side? I am not really sure where to start.
Brent

That explains your problem.
Your AirPort Express (AX) can't support both Ethernet and wireless clients because it only has a single Ethernet port. Possible solutions:
Get an inexpensive wired router. Connect your cable modem to the WAN/Internet side of the router. Connect your PC and AX to the LAN side of the router. Configure the AX so that the option to distribute IP addresses is DISABLED.
OR...
Replace the AX with an inexpensive wireless router or even the AirPort Extreme base station (AEBS). These will support both Ethernet and wireless clients.

Installing IPM version 2.4

We currently have a Installation of Ciscoworks Lan that was upgraded to all of the latestest versions(RME 3.5, Common Services SP2, DFM 1.2, CM 3.3) as well as windows 2000 SP4 and all current windows updates. Also all of the Apache and SSL updates were done. We also have the WAN management package and want to use the IMP package to be able to monitor a new VOIP deployment that will be going live in the upcoming weeks. I have some questions on the installation
1) It appears that the LAN and WAN solutions share the same common parts(RME, Common Services and CiscoView). Can I just install the IPM product using what is currently installed or do I need components from the WAN package as a prerequeset?
2) If I can install, can I use the media I have for 2.4 and upgrade to 2.5 or is it better to just download and install 2.5?
3) Are these versions of IPM compatible with win2k SP4?
4) I saw a caveat about not being able to use the SSL with 2.4 but it appears to be ok with 2.5. I need clarification on that.
Thanks in advance.

IPM 2.4 cannot be installed with the new Common Services 2.2 bundle. Cannot mix n match old versions with new ones. IPM do also do not require any other application like RME, Campus etc for installation either. Going with IPM 2.5 is a better option. For that, you just need to first install Common Services 2.2 and then IPM 2.5. Though you can still install IPM 2.5 on the same server which have RME 3.5, Campus 3.3 etc. But you can also install IPM 2.5 on a standalone server with Common Services 2.2 installed first. See Software Requirements: http://www.cisco.com/univercd/cc/td/doc/product/rtrmgmt/ipmcw2k/cipm25/ig/winnt.htm for details.

SF300-24P VLAN Confusion - autosmartport not being too smart?

Hi Everyone, first question i've posted, i'll try and give as much information as possible, i'm an extremely quick learner as well and have been around networking for nearly 20 years but this is my first outing into the medium sized VoIP deployment with prioritised LAN traffic and a client that is itching to say "told you so" about using IP phones.
I have 4 x SF300-24P switches in a network i'm deploying, 1 will be adjacent to the router (a draytek Vigor 3200 - 4xWAN Gigabit) and the other 3 will be trunked using the GE/01-GE/03 ports to the main switch and will then distribute through a patch panel to give me 96 network ports with PoE capability where required. There will be 30+ IP Phones on the network, all of which are Yealink T38G SIP handsets.
I want to have two VLAN's - one for regular workstations, and one for IP Phones with the IP Phone VLAN getting high prority for its traffic on the LAN - all documentation makes it sound simple but it doesn't seem to be working the way I think I expect it to. I don't mind the two VLAN's sharing the same IP address space at this time and currently all occupy 10.0.0.0/24 internally.
So, I have 2 questions and a problem.
First, from the factory, the switches are configured that VLAN1 is the default VLAN and that auto-voice VLAN is also VLAN1? Is this right?
Second, i'm having trouble determining the difference in terminology for port types between general, access, trunk etc - obviously trunk is between switches and carries VLAN information through to the next segment of the network.
My main problem seems to be with auto-voice VLAN and smartport. If I enable smartport, the switch figures out through LLDP that the port is used by an IP Phone + Desktop (excellent, this is what I want it to do) so then puts the handsets in VLAN1 but then the handsets start to become invisible on the network after 2-3 minutes, the handsets then reboot because they've detected a network drop out and then reconnect, re-register at the voice server and are visible and contactable for 2-3 minutes then the loop begins again.
If I disable smartport, the problem goes away.
Am I unreasonably expecting that any user can unpack an IP phone and (subject to provisioning on the server), plug it into any port on the network and it will figure out that it's a phone, not a PC and then prioritise its traffic?
What I want to avoid is the possibility of internal bandwidth lag if someone copies a large file over the network and people are using the phones that the phone users don't get packet loss or audio instability because of the file copy. The internet side will be fine, the Vigor3200 has QoS facilities built in and i've had good success on smaller networks with these routers.
Ideally I need a semi-planned network setup where people with WiFi SIP clients will also get some priority.
I have set QoS on the handsets to match DSCP46 from the switches - can the traffic be manipulated this way or does it already do that in the DSCP to Queue setup which automatically puts anything above 40 in Queue 4 (high priority).
All help very greatfully received.
James

Hello James,
Welcome to the forums!
About the default settings. The switch comes with vlan1 as the default vlan for all traffic.
Here is a quick overview of the port settings
access - one vlan
trunk - multiple vlans
general - multiple vlans (had additional options)
When using the auto voice-vlan, you can have your port set as access for vlan 1 and when the switch see a phone connected, it will join the voice vlan also. This allow the ports to be dynamic. It is not necessary to do this. You can create all ports as trunk ports that are part of both your default vlan and your voice vlan.
The benifits of auto-voice vlan
-phones are discovered and joined to the vlan dynamically
-predetermined QoS settings
-security in that you can have your port set to access
This is a relatively basic overview.
As for the problem you are seeing. I would recommend that you check the firmware of the switch and upgrade if needed. While it may not have anything to do with the problem at hand, it will help prevent any future issues.
I would suggest disabling the Green Ethernet, which can be found under the port management section. If you continue to see the problem after that, I would recommend giving us a call at the support center. We will be able to look a little closer to what is happening.
http://www.cisco.com/en/US/support/tsd_cisco_small_business_support_center_contacts.html

Call forwarded Caller ID

Hi Folks,
I have an issue in a new Voip deployment whereby when a user call forwards their internal phone to an external mobile, the site prefix is being added to the Originating caller ID on the destination mobile phone and trailing digits are stripped.
For example..
the site prefix is +43166105
A user outside the company with mobile phone number 06991968XXXX calls an internal extension that is call forwarded to 0664545XXXX, when the call rings on 06664545XXX, the caller ID is shown as +431661056991968, with last 4 digits missing
The site has two voice gateways, GW2 for specific mobile phone calls, and GW1 for regular calls and all other mobile calls, when a user call forwards to a mobile phone that is reachable via GW2, the caller ID is showing correctly (both g/w are connected to different providers)
I have a translation pattern on both g/w that strips the leading zero's of the calling number and also makes sure call type is presented properly to the provider.
Generally, i would be calling the provider, however the site are telling me that this worked OK on the previous PBX based setup.
Here are some relevant pieces of configuration information, and the ISDN debug of an affected call, as you can see, we are providing the correct digits to the provider, yet we always see the prefix regardless
I have tried changing the numbering type in the translation pattern from any to unknown and it has the effect of then showing no digits of the Caller ID, it will just show the prefix number.
508090: Mar 28 10:42:47.084 CET+1: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x07CC
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98382
Exclusive, Channel 2
Calling Party Number i = 0x0181, '6991968XXXX'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '0664545XXXX'
Plan:ISDN, Type:Unknown
GW1
trunk group A1_Telekom
carrier-id Vienna_A1_Telekom
description Vienna_PSTN_Trunks
hunt-scheme round-robin both
isdn supp-service tbct notify-on-clear
translation-profile outgoing OUTBOUND
voice translation-rule 4
rule 1 /^01$.*$/ /01\1/
rule 2 /^0/ //
rule 3 /^8912$/ /0017275788912/
voice translation-rule 5
rule 1 /\+43166105/ /166105/ type any national
rule 2 /^000$.*$/ /\1/ type any international
rule 3 /^00$.*$/ /\1/ type any national
voice translation-profile OUTBOUND
translate calling 5
translate called 4
dial-peer voice 99 pots
trunkgroup A1_Telekom
service session
destination-pattern 0T
fax rate disable
direct-inward-dial
forward-digits all

Hi Steven.
Can you please activate a debug ccapi inout on your VG1 and make a call?
After that please attach the output here.
Thanks
Regards
Carlo

PoE Problem with SRW224P

Good morning everyone,
I have 15 of these switches used for a Voip deployment.
At one point, there was a power failure on one of these, and when it turned back on, the phones didn't power up.
The switch was fully operational.
As I logged in to the switch to check PoE port config, everything was enabled, yet everything was still off, including the PoE lights on the switch. When I clicked submit on that screen, everything powered on.
This happened at least 3 times, and Linksys suggested we change the switch.
Now, this happened on another switch. We tried many different firmware versions, and we always have the same problem.
Has anyone experienced something like that?

Got the exact same thing, I thought the VoIP phones were the issue, but I got new phones just today and those don't work either.
So the switch's POE just died after a power failure...
Mike

DiffServ & TE paths are required to deploy commercial VoIP over a MPLS net.

Dear members.
There's something I'd like to know from those that truly and successfully
implement commercial VoIP over a MPLS infrastructure.
Supporting IP QoS (DiffServ) across the core backbone and/or MPLS traffic-
engineered paths is really required ?
Consider a whole backbone built with high speed throughput connections, plenty of bandwidth availability and no periods of congestion!
I do believe TE tunnels (paths) are necessary for quick recovery when a node
or link fails, but rather them assuming something I haven't experienced yet
I'd like to listen from those who really have already successfully deployed
commercial VoIP.
Best regards.
Murilo Pugliese.

With increasing adoption of voice over IP (VoIP), the landscape for deployment is rapidly changing. Service providers are often driven by the need to provide customers a high grade of service to carry voice traffic across a network. However, today's multiservice packet networks rely on IP-based packet switching. In addition, IP by itself is simply best-effort service that is not sufficient to provide the strict delay, jitter, and bandwidth guarantees required for VoIP and other real-time traffic. Cisco IOS QoS features are ideal for this situation. Using the IETF differentiated services (DiffServ) model for QoS, VoIP traffic can be treated appropriately.
http://www.cisco.com/en/US/netsol/ns341/ns396/ns172/ns155/networking_solutions_white_paper09186a00800a8441.shtml

Wireless VoIP roaming - getting an 802.11g phone to roam with CCX / CCKM

Hello,
Our group is facing a pretty difficult issue at the moment, we are trying to deploy a wireless Avaya VoIP phone at one of our sites in Australia.
The site in question is located in Western Australia, which is on the other side of the continent to where our 5508 WLC is (in Sydney), so there is a latency of abouy 60-80ms between the sites. When testing the phone at the Sydney location, we're able to authenticate the phone via a number of methods (802.11x, 802.11x FT) and the phone can roam fine. When testing at the WA location, the 802.11x authentication method drops the call when roaming, and the 802.11x FT authentication method has about 2 seconds of dead air during roaming / reassociation.
The avaya phone supports a few different methods of "fast handoff" - CCKM and OKC - and we've only got the OKC (802.11 version of fast transition) method to work, not CCKM. The phone reports error messages when trying to connect to the network with CCKM enabled - specifically "No CCKM AP's" and "No CCX AP's".
So I have some questions...
1) Is a 60ms latency between sites a dealbreaker for wireless voice usually?
2) The voice engineer is convinced that CCX is disabled on the WLAN. My understanding of CCX (according to online Cisco docco) is that it can't be switched off. Is this the case? I know CCKM can be disabled (in the Layer 2 Authentication options) but can CCX or CCKM be enabled / disabled anywhere else?
3) The voice engineer has a point on one front - if we do a command "show ap ccx rm [APName] status", the command outputs "Beacon Request ..... Disabled." This seems to be related to CCX Location awareness - but even after switching this on in the WLC (both globally and for the WLAN), I cannot get this to enable.
3) We have had some minimal success getting the phone to connect with only CCKM Authentication enabled. The phone could not roam. When we did this, I looked at the Client information in the WLC of this device and the CCX Version read "Not Supported." Does this perhaps mean some kind of CCX compatibility problem?
4) Is there much of a difference between OKC (Fast Transition) and CCKM in terms of roaming time? I.e. are we better off trying to install a WLC at the location rather than getting this work remotely.
We're a bit lost on this. Anyone that's had experience in getting Wireless VoIP phones to roam on a wireless campus before, particularly where the WLC is remote, I'd love to hear from you.
Thanks,
Dave

Hi, thanks Steve. Aironet IE is enabled.
The encryption is news to me. We intially tried WPA2/AES & WPA/TKIP together but had no success at all. We then moved to WPA/TKIP, and we were able to get the phone to connect to the WLAN using the CCKM settings, but it doesn't roam at all & the WLC reported the Clients CCX version was "Not Supported".
I think the phone has CCX version 4. Are you certain that the phone will only support WPA & WPA2 with CCX v5? Just that this sounds a little strange, the phone's documentation claims to support CCKM, but if you're right then there's no way we can secure our phone conversations? In previous testing, we did actually temporarily drop all authentication / encryption and the roaming worked perfectly.
Thanks for your help...

Wireless VoIP Across Subnets (Cores)

We have a campus with 3 buildings
Building one and two is internally routed with a L3 switch (6509) 10.2.0.0/16
Building three is routed with a L3 switch (4900) 10.3.0.0/16
Our VoIP network in building one and two is 10.2.11.0/24
Our VoIP network in building one and two is 10.3.111.0/24
Our wireless system is a Meraki-based system that we just implemented.
The issue is that the wireless completely drops when moving from one subnet to the other. So in effect, service is pretty good when going between building one and two (share the same core and subnet), but drops when going into building 3. Signal to AP is just fine.
What do I need to look at and/or do to get prevents drops when roaming between the two subnets (cores) ?

Do not have any knowledge about Meraki wireless system. But as per your problem discription mobility/roaming is not working in your network.
In a normal cisco WLC environment client will not change its IP when roaming, in order to provide smooth roaming experience. Traffic will tunnel back to original controller where they first associate to network.
You need to bring this to the people (hope this was not deploy by yourself) who deploy this and ask them to fix it
Rasika

Sending an INVITE to my VoIP account by a SIP Servlet

I tried to make an outbound call to an ordinary landline number through my VoIP account by a SIP Servlet, but never been successful.
I tried something similar to the following:
javax.servlet.sip.SipFactory factory=(javax.servlet.sip.SipFactory) this.getServletContext().getAttribute(javax.servlet.sip.SipServlet.SIP_FACTORY);
javax.servlet.sip.SipApplicationSession appSession =factory.createApplicationSession();
to = “sip:land_line_number@my_voip_proxy_server.com”
from=”sip:my_voip_did_number:my_voip_password@ my_voip_proxy_server.com”
javax.servlet.sip.SipServletRequest invite = factory.createRequest(appSession, "INVITE", from, to);
invite.pushRoute(URI of my_voip_outbound_proxy_server.com);
invite.send();
I use Weblogic SIP server. I configured sip.xml, and everything else is correct to my knowledge.
I always get the error message something similar to “Cannot resolve path to my_voip_proxy_server.com”
(Note: When I use SJPhone softphone with my VoIP account details, it simply rings to a landline)
Is the way I incorporating my_voip_outbound_proxy_server correct?
Any help from anyone is greatly appreciated.
Thank you very much.

hi there,
I am at the same point too..
I am using a softphone called mirrial , Using Bea WEB Logic SIP Server...Deployed sar arhieve , did not define any outbound proxy in client side , did not specify default servlet name for sip server..
Is there anybody who succedded such kind of proxy calls ?
And can clarify the details if have?
Thanks

Txt file missing after deploying war

All,
i've encountered weird problem when i'm deploy new war to the server. i want my class read the txt file. so that i've put the txt file at the same path as class as well at src/javapackage. here is the path..
F:/myapp/voip/src/clientpkg
so that i assumed when i deploy the war file to the server all classess & txt file will be kept at the ...
webapps/voip/WEB-INF/classes/clientpkg/
but when i checked at the clientpkg directory, the txt file is missing. what shud i do now?
here is my code to read the txt file..
private boolean migration25Cpes() throws Exception {
boolean status = false;
try {
System.out.println(System.getProperty("user.dir"));
System.out.println("read the file again/");
InputStream in = this.getClass().getResourceAsStream("list25cpe.txt");
Reader rin = new InputStreamReader(in);
BufferedReader inputFile = new BufferedReader(rin);
String readSource = "";
while ( (readSource = inputFile.readLine()) != null) {
System.out.println(readSource);
System.out.println("the input now : " +inputFile.read());
inputFile.close();
catch (FileNotFoundException e) {
e.printStackTrace();
return status;
}

Then you have to find a way to tell JBuilder to include the text file in the jar. I don't know JBuilder, but I'm guessing that right now it is probably building the jar from the compiler output directory, which contains only class files.
There may be a setting to copy non-java files to the compiler output directory, or you may need to put stuff you want to appear in WEB-INF/classes in some other location for it to be picked up by the JBuilder war-creator.

VoIP Deployment

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