VoIP network

Hi, I'm new here. I have UC 560 FXO, 2 Cisco 2900 Router,2 Cisco Catalyst 3560 V2 series PoE-48 and 4 tel Cisco 7942. I want to create a VoIP network from point A to point B ; where I can find all settings, connections and configuration commands for these?
Thank you,
Adrian.

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Just to clarify, when you write of running two cables to each cubicle, one for data and one for VoIP, the two sets of traffic never share a link (or device)? For example, both those links wouldn't terminate on the same device?
If any part of the network infrastructure shares physical resources with both the data and VoIP traffic, there's a chance the data traffic might be adverse to VoIP. If there's any physical resource sharing, QoS might be configured to guarantee resources to VoIP traffic over data traffic.
As LANs often have ample bandwidth, and it's often easy to provide more bandwidth, if necessary, you might not need to utilize QoS to have acceptable VoIP.
PS:
BTW, older hosts used to have TCP stacks that would cap their bandwidth utilization on high speed LANs. Newer TCP host stacks can often drive even 10g host links at full utilization. QoS might be needed where it didn't used to be needed.

Similar Messages

Constant noise in background in voip network

Hi,
we have 3 branches and 20 small home offices setup.
we installed CUCM 8.6 running and serving all the branches and home offices.
All branch voip network is good, but we hear background noise while calling home office ip phones, the background noise is heard at headoffice only, at home office side it is good.
i couldn't figure out where it is going wrong.
the setup is like
IP Phone------> call manager------> gateway(cisco 1940)------->internet----------> homeoffice router(cisco881)--------->ip phone.
the home offices are connected over DMVPN tunnels. and the bandwidth is 1 MB for the tunnels.
attaching router config for head office gateway and home office router

Hi Abhijit,
thank you for the reply,
its some kind of crakling sound.
we are using 6921 at home office side and 8941 and 6921 at headoffice side, but i found no issue with phones,these are working fine in internal network, when i did .mute the phone at home office side, there is no sound on head office side..
is there any problem with the encryption used on tunnel, i applied qos for the voip but found no difference..
please can you help me...

GSM to VoIP to GSM network

Hi guys,
I would like to ask if somebody knows how it is possible to make the responder site of the GSM network(a mobile phone) to be able to see the caller ID of the caller(mobile phone), because in the meanwhile the call have to pass through the VoIP network and so at the end, in the BTS it has to use other SIM card again.
Thanks a lot!

Chris:
I think it is better to put your question in voice forum.
Amjad

Time different on VOIP and data network

i have a 5 minute time difference on the data network than the VOIP network.
i started time services on the Pub and changed the router config to use an internal server that synchs with the clock in Boulder.
any ideas?
thanks!

Jerry,
Here is a link for tine service with some recommendations. However I would not use times as when you do an upgrade to CCM or install an SR XNTP will be installed & you will have t time sources fighting for time. Having 2 time clients fighting of the correct time on the servers is a BAD thing!!! I would use XNTP & point all of the Subs to the Pub & point the Pub to the same source that is being utilized by the data network. Please keep in mind that every time you install an SR or upgrade the XNTP config file for all servers will be over written with the default on all nodes, thus you will need to make the appropriate changes after to keep your time in sync.:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_white_paper0900aecd8037fdb5.shtml
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008009470f.shtml
Mike

Migrating from SonicWALL to RRAS - questions on functionality for business network setup & monitoring

At my organization we are planning to move away from using SonicWALL and using Microsoft Server 2012 RRAS to handle IPv4 NAT routing and VPN connections for remote employees. I am trying to plan for all the functionality I currently have in our
SonicWALL router, and I'm not familiar with the Windows equivalent for some features and/or if the available features will have all the functionality our organization needs. Additionally wondering if there is something better that you would recommend
in terms of ease-of-use / management from an IT standpoint.
Feature: Firewall
I assume that Windows Advanced Firewall can handle all the firewall settings that I would normally find in an enterprise router. Is this a correct assumption?
Feature: Content blocking
What Microsoft technology would I use for blocking certain content and websites from being viewed in the workplace network? Can Windows Firewall handle these two types of content blocking?:
Blocking websites by general content - i.e. block porn sites.
Blocking websites by specific URL
Feature: Network activity monitoring
In cases where the network is slow, what would I use to inspect what IP/MAC is hogging bandwidth? Currently on SonicWALL we are able to see which computer is hogging bandwidth, see what website is, how much bandwidth it is consuming. Can RRAS
handle this level of detail, or is SCOM / some other technology more well-suited? Should we use a non-Microsoft application (something like Wireshark) instead?
Feature: Network prioritization (QoS / bandwidth management)
Can we set certain network groups to have priority over others? For example, if we split our incoming network into two networks using RRAS, 192.168.50.x for data and 10.0.50.x for VOIP, I want the VOIP network to get priority in order to
ensure quality audio on all phone calls. I assume this is available in RRAS.
Basically I'm just looking to see what systems I need set up to satisfy these business needs. If anyone has any real-world experience let me know what your setup is like -- I'm curious to know!

"I assume that Windows Advanced Firewall can handle all the firewall settings that I would normally find in an enterprise router. Is this a correct assumption?"
Your statement as it stands is correct. In fact, Windows firewall will handle more firewall settings than those found in a
router. However, SonicWall is not just a router - it's a firewall. There is a big difference. Given that you are comparing Windows Firewall to a robust firewall, I agree with Dennis that you should either stay with SonicWall or
look at Microsoft's product. But, given that on the link Dennis provided, you will find that Microsoft is discontinuing UAG, you might want to stick with what you know - SonicWall. Or, you need to evaluate your needs more. Yes, Microsoft
provides a number of built-in capabilities that may address your particular needs. We can't answer that. But, if there are features you need that are not available via Windows Server, then it is best to get a product that addresses those needs.
.:|:.:|:. tim

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

Small office VoIP system

Good morning everyone,
I am currently defining the architecture for a very simple VoIP network that is to be installed in a small office (about 70 VoIP extensions). Initially, we don't want to include any special feature, just the internal voice IP service and the ability to make up to 4 simultaneous external phone calls through PSTN.
We are going to acquire a Cisco 2921 Router, the SL-29-UC-K9 Unified Communications licence and a VIC2-4FXO.
I have been reading about ISR G2 licensing, but I am still not sure of fully understanding what we need for this project. My question is, does the UC licence for the Cisco 2921 include the FL-CME licence to use the Call Manager Express functionality? do we need something else? any special license for the FXOs functionality?
Thank you for your help
Pablo

Thank you guys for your answers,
Actually, it's my first contact as a developer with VoIP and UC Cisco technology... so I would like to keep it simple at the beginning: just internal voip phone calls and the possibility to ring pstn extensions. Our budget is very limited too... so I need to stick to the hardware that I told you in the previous message: 2921 + fxo card.
As far as I have read, SRST functionality is very useful when you have several nodes, to provide survival to each local node that loses connectivity with a centralized call manager. But in our scenario, with a single node... I think we can give up this functionality.
So, we only have budget to purchase any other license that could be necessary to accomplish this small project. so the key question is: does SL-29-UC-K9 Unified Communications License include Call Manager Express functionality and the ability to use de FXO card to make pstn phone calls as well? because that will be more than enough for us...
thank you very much
Pablo

VOIP calls not being forwarded from GW to GK and visa versa

I have a VOIP network setup in place using 2691 routers. i have a siemens PBX connected to the 2691 routers using an E1 controller (running QSIG), my WAN interface between the two sides is an ATM interface. when i do a csim start from one router to the extension on the other side the call goes through and the phone rings. However, when i actually dial from one extension to the other, the call is not being forwarded. and what happens is that i hear the dial tone again on the phone im dialing from.
I have taken some debugs but still could pin point the problem, even though in the ccapi inout debug i found that when it tries to look for a node, it doesnt find it.
I have attached the configuration for both sides. Along with the debug from one of the sites.
There are a couple of GW's associated with the GK.
The ones im concerned about are Mussalla_GW & GK-HO.
Thank You in advance
Mouhannad

Try this:
dial-peer voice 2 pots
direct-inward-dial
Without this command the router will not collect the digits coming across the PRI.
Please rate helpful posts.

Ip SLA RTP based VOIP Operation - To find out MOS value

Hi All,
I am new to VOIP. We are trying to find out the MOS value in our VOIP network. For that we thought of using IP SLA RTP Based VOIP operation to get the MOS values. http://www.cisco.com/en/US/docs/ios/12_4t/12_4t4/htrtpvip.html
I ve used 3825 with NM HDV module with 3 DSP as SLA originator and AS 5400 XM as SLA responder.
But i'm not getting the MOS values,
show ip sla statistics shows that the operation failed due to Format Failure.
I ve attached the config of my 3825. Kindly go through it and advise if any changes to be done.
In AS 5400 XM there is no special config related to this. I ve enabled only " IP SLA RESPONDER"
Error message:
LAB-3825-R6# sh ip sla stat
Round Trip Time (RTT) for Index 1
Type of operation: rtp
Latest operation start time: *05:04:58.707 UTC Wed May 14 2008
Latest operation return code: Format failure
Latest RTT (milliseconds): 0
Source to Destination Path Measurements:
Interarrival Jitter: 0
Packets Sent: 0
Packets Lost: 0
Estimated R-factor: 0 MOS-CQ: 0.00
Destination to Source Path Measurements:
Interarrival Jitter: 0
Packets Sent: 0
Packets Lost: 0
Estimated R-factor: 0 MOS-CQ: 0.00
Operation time to live: Forever
Operational state of entry: Active
Last time this entry was reset: Never
LAB-3825-R6# sh ip sla stat aggre
Round Trip Time (RTT) for Index 1
Type of operation: rtp
Start Time Index: *05:06:21.019 UTC Wed May 14 2008
Number of successful operations: 0
Number of operations over threshold: 0
Number of failed operations due to a Timeout: 0
Number of failed operations due to a No Connection: 1
Number of failed operations due to an Internal Error: 5
Number of failed operations due to a Sequence Error: 0
RTT (avg/min/max): 0/0/0 ms
Source to Destination Path Measurements:
Interarrival Jitter (avg/min/max): 0/0/0
Packets Sent (avg/min/max): 0/0/0
Packets Lost (avg/min/max): 0/0/0
Estimated R-factor (avg/min/max): 0/0/0
MOS-CQ (avg/min/max): 0.00/0.00/0.00
Destination to Source Path Measurements:
Interarrival Jitter (avg/min/max): 0/0/0
Packets Sent (avg/min/max): 0/0/0
Packets Lost (avg/min/max): 0/0/0
Estimated R-factor (avg/min/max): 0/0/0
MOS-CQ (avg/min/max): 0.00/0.00/0.00
Any help is greatly appreciated.
thanks in advance.

Hi,
AS 5400 cannot be used even as SLA responder for RTP probe. Thats the reason i got the Format Failure error. We can view the type of SLA Probes the router supports by issuing the following command:
sh ip sla application.
for eg below is what i ve taken from AS 5400
sh ip sla application
IP Service Level Agreements
Version: Round Trip Time MIB 2.2.0, Infrastructure Engine-II
Time of last change in whole IP SLAs: 10:48:00.737 IST Tue May 20 2008
Estimated system max number of entries: 49625
Estimated number of configurable operations: 49608
Number of Entries configured : 17
Number of active Entries : 17
Number of pending Entries : 0
Number of inactive Entries : 0
Supported Operation Types
Type of Operation to Perform: dhcp
Type of Operation to Perform: dlsw
Type of Operation to Perform: dns
Type of Operation to Perform: echo
Type of Operation to Perform: frameRelay
Type of Operation to Perform: ftp
Type of Operation to Perform: http
Type of Operation to Perform: icmpJitter
Type of Operation to Perform: jitter
Type of Operation to Perform: pathEcho
Type of Operation to Perform: pathJitter
Type of Operation to Perform: tcpConnect
Type of Operation to Perform: udpEcho
Type of Operation to Perform: voip
IP SLAs low memory water mark: 68416281
chnmgw1#
Hope this will help others looking for RTP based VOIP operation..

Call info from VOIP Router for billing purposes

Hi Pros,
My client has a VOIP network where all routers have FXS Cards & Dial Peers configured
--- No CallManager ---
They want to bill calls using router outputs.
Now, I thought of collecting debugging data on syslog servers. Can anybody help identify a debug command which can generate call setup (calling party & called party) & time info?
Pratik

Hi,
Check out this link.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094e72.shtml
You can use syslog. You would need some clever parsing to generate the billing etc i think though.
Cheers,
Tim.

Voip use ivr to reply instead of busy tone

All
The T1 ISDN got 300 number on a IDAP, and some of the number does not use.
The current network is Cisco 2600 T1 gw + 3600 gk.
The voip network had been formed.
Now if the number not need in use, it just simply give out a busy tone...
but is it possible for me to play an audio "this number is not yet register with our company...."
how can I do it? any document refer to this.
got a basic knowledge on tftp audio file...

Yes, it is possible. You can create a IVR application to let the user know that a number does not exist or it is not yet registered to the company. Check the configuration for the same in the following document :
VoIP with IVR
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_example09186a0080094305.shtml

Unified Communications Network Management Licensing

Hi,
i have purchased all 4 VOIP Network Management Software (CUOM, CUSM, ecc...) with their PAK. Now i'm going to register the 4 PAK, but i have a doubt: Cisco requires the MAC address of the server that hosts the Management Software. I have installed the Cisco software on 2 HP dl380 servers with 4 NIC's and, on each server, i configured 2 NIC's in load balance (with embedded HP network software): the other 2 NIC's are disconnected.
Is there someone that can tell me which MAC address i must provide to Cisco PAK registration ?
Thank you,
Roberto

Interesting question. When you configured the network team, the MAC address of one of the physical interfaces was grabbed and assigned as the team's primary layer 2 address. If I recall correctly, this would normally be the first (lowest numbered) ethernet interface. But it is likely more complicated than that. But, the point is I would lean to using the mac address reported for the team by the network configuration utility (NCU).
A good doc/pdf on the topic: http://h20000.www2.hp.com/bc/docs/support/SupportManual/c01415139%2Fc01415139.pdf
HTH.
Regards,
Bill
Please remember to rate helpful posts.

IPhone and Business VOIP? Can this happen?

I know this may be a long shot but I cannot be the only one wondering about if this can be done. I have been searching for the possibility of integrating my recently purchased iPhone into my work VOIP network. I have come across a few VOIP applications for the iPhone but have not found any information that can confirm this possibility. We use Cisco CallManager (version 5) at my work and I am one of the IT staff so I know I can do this if I could get some information that can state if it can be done in the first place. Cisco CallManager has the ability to work with SIP devices so if the right iPhone application is out there I am hoping I can make this work. If anyone has tried this or found some useful information I would be interested in reading it.

First keep in mind that ATT and Apple will only allow for VOIP over WiFi, not over Cellular (3G, Edge, etc.)
If you search the App store for "Fring" you will find an IM client that also allows for Skype and SIP connectivity. You will have to set up free Fring account (I think). You can go to http://www.fring.com/ to find out more.
I have tried it with WiFi using a free SIP account from voipstunt and it does work, the only drawback tends to be not getting dialer after connected (in case you want to push digits to join a conference or something), but I have not really played with it that much.

N95 SIP/VOIP Calling

Hi everyone,
I have just recently bought a Nokia N95 and was trying to configure the SIP setting on it to work on my VOIP network. Now for the life of me i can't seem to get it to work it is currently on T-Mobile and just wanted to know if anyone has successfully got SIP working on a N95. If you have do you mind posting an example of your settings please so i can have a look.
Thanks
Rockyuk

I recommend the following SIP Providers:
Sipgate
VoipTalk
Both SIP provider's websites have instuctions how to configure the earlier Nokia phones, and these instructions also work for the N95. I plan to write up a how-to guide for the N95, which I will publish on my website, but this won't be ready for a week or so.
Sipgate is probably slightly better as you get a free answerphone and a free 0845 or local number assigned to your SIP account.
I live in France, where unmetered GPRS access is expensive. So I only use these SIP services at WiFi hotspots at home and in my hotel.
However, a friend of mine has T-Mobile. He has unlimited tarif from T-Mobile. He has configured these two SIP providers on his T-Mobile branded N95, and they work extremely well on the 3G connection.
You are lucky to have a T-Mobile service. As I am sure you are aware Orange and Voda have removed the SIP client from their branded N95.
Someone correct me if I am wrong, but I think Orange and Voda have also blocked the SIP ports (typically, UDP 5060 - UDP 5065). So even if you had a generic N95 you would not be able to use the SIP client with Orange or Voda !
Rob Wisbey
Freelance Technical Author | Useful Nokia N95 Information

VoIP and VRFs

Does anyone know of any concerns, issues, problems, or hidden gotchas that have been experienced with creating a VRF for a VoIP network? What I would actually like to do is place everything (except the media gateways) in a VRF and firewall it. Thus only call signaling, management traffic, and any required database connectivity would have to pass through the firewall. Any thoughts, anyone?

Firewalling voice is always a headache. Unfortunately a lot of signaling protocols are proprietary like SCCP, and MGCP (not really). Or just change a lot, or not completely standardized like SIP.
Between the time a Dev on a VTG group decides to add a new field to a protocol like SCCP, and the time it takes the corresponding Dev on a Firewall group to add the support for that field on its 'Inspection' engine sometimes takes months. And the fact that all communications are opened on random dynamic ports between the 16K and 32K makes matters worst.
I do think it's a good idea, specially with cybersecuirty threads on the rise, and toll fraud so prevalent this days. I think SBC and Media relay points are a good way to get everything more in control.
I just wanted to raise some awareness that if you want to go down that path, you do need a solid roll-out and testing plan as things will likely get bizarre a few times.

VoIP network

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