Media Endpoint Discovery is an enhancement of LLDP, known as LLDP-MED, that provides the following facilities:
Auto-discovery of LAN policies (such as VLAN, Layer 2 Priority and Differentiated services (Diffserv) settings) enabling plug and play networking.
Device location discovery to allow creation of location databases and, in the case of Voice over ...
You are correct this is a VoIP configuration. This is part of the configuration for CVP call center survivability. If this site lost it's connection to the Internet or WAN (however its connecting back to your Data Center), the 2800 would take over call processing for the IP phones. It is used in combination with the standard SRST functionality on Cisco'...
Depending on which version of 12.x you're on, you should be able to do the following:
router(config-sip-ua)#no transport tcp
router(config-sip-ua)#no transport udp
This stops the router from listening on port 5060.
The term you are looking for to describe this behavior, is an Outbound Campaign.
There is no built in functionality in Cisco Call Manager (CUCM) itself to do this.
However, both of Cisco's Contact Center products do have this ability. Unified Contact Center Express (UCCX) has a more limited capability for performing a Campaign, but is a much cheaper, much ...
Your question's pretty broad. There's a lot of different commands you can use to troubleshoot and monitor QoS, so I'll focus on the primary question you have, which is how to reasonably verify your QoS configuration is working and how to read the policy-map interface output.
The only true way to verify that QoS is working is to hook up a traffic generator ...
Here you go!
1. Enter RTP in the display filter in Wireshark when the capture is open.
2. Now select an RTP packet in any stream and click on the menu option Telephony.
3. Go to RTP>Stream Analysis and click on Save Payload button.
4. Change the format to .au and choose a name (lets say test.au) and location for your file.
5. Play it using your favorite ...
Because in some VoIP protocols signalling (setup of connection, but also DTMF tones) can be carried in separate session from voice payload. If the voice payload can't traverse both ways (and it's often the case with NAT/firewall gateways) you'll get one way audio, but the connection still will be made and can provide some additional services (like your tones)...
VoIP over the public Internet can be a problem, but it usually works good enough, most of the time, although there can be times where it sucks. Most ISPs have extra cost features where they will honor some of your QoS markings and policies.
(I know Verizon Business, among others, has some specific packages for QoS, and you may need to adjust your policies ...
Link Layer Discovery (IEEE-802.1AB-2005) provides the base capabilities for discovery, but was seen as insufficent for IP Telephony and Real Time Video. So the LLDP-MED initiative was formed by HP, Mitel and Avaya in 2004. What does LLDP-MED provide?
Interoperability > provide vendor independent management capabilities, enabling different convergence ...
It's the PSTN part that cost money.
You have to interconnect with the regular phone network and depending on the provider the costs vary (fixed cost per month or per call, etc...). Using IP to IP calls you do not connect to the PSTN network, hence no need to pay for that part of the infrastructure
Is QoS an issue for VOIP calls over the Internet?
Sometimes. The answer why is related to the extinction of these objects:
(for people born after the year 2000, that's a phone booth)
QoS is only an issue if the customer thinks it's important enough to pay for.
it's important to maintain QoS on an internal network (...
Do you need QOS on a dedicated VoIP network?
Possibly not, but you probably want it in place.
It really will depend on the network layout/capacity, the complexity of your phone system, the volume of voice traffic, and your business needs.
Here is the most basic example I can think of:
If a link is so saturated with calls that call-control traffic can'...
You're buying SIP Trunking from a provider, so that you don't have to do all the work yourself of connecting to the global telephone network.
If you wanted to "roll your own" SIP trunking service connected to the PSTN, you certainly could. It would just be extremely cost prohibitive if that was not what you were in business to do. In 99% of cases it is ...
First, you should probably not use VLAN 1 - it's generally not recommended, or if used, used for network device management traffic only. But, to answer your question, yes, you'd be using inter-vlan routing on the Cisco switch. See config below:
! enable layer-3 routing
! declare vlans
name my vlan 1
name my vlan 21
Should TCP slow start be used with SIP-over-TCP?
SIP does not transfer large data with TCP. SIP is only used for signaling, the transport of the actual data (voice, video, ...) is done with RTP and thus using UDP. The SIP messages are small enough to not be affected by slow start. The control flow of SIP is not affected either by slow start since messages ...
ASA's can preserve DSCP and match traffic based on the DSCP markings, but it does not have the ability to mark/change DSCP.
Your best bet would be to place a switch (or other device) that can mark DSCP before the ASA to mark your traffic.
However, I do want to note that if the "public" network on the ASA is to a service provider (as is often the case) that ...
If your 2811 is providing the phone services for your Cisco IP Phones at this site, finding the IP Address of the 2811 is relatively easy:
Press the "settings" button on one of the phones
Choose "Device Configuration"
Choose "Unifed CM Configuration"
Choose "Unified CM 1" or whichever says "Active"
The listed IP address is the IP address of whatever device ...
Now, I've been told that routers with inbound QoS policies are capable of leveraging the TCP congestion avoidance mechanism by dropping ACKs in order to rate-shape inbound traffic. I've never seen this done in practice, or at least never seen it work; is this a real thing? If so, is it effective? Wouldn't it result in a run of dropped packets every time that ...
VoIP runs over IP, so it is independent of layer-1. The only sort of restriction based on layer-1 would be the bandwidth required. Voice is not separate from data on the wire.
You may be meaning a VoIP phone, which may be powered by PoE, but most have wall-warts, too. If the Category-3 cable has all four pairs, there is no real reason the it couldn't ...
By themselves, VLANs do not offer any security. But you use them to implement layer 3 security.
Remember that VLANs are layer 2 concepts, while IP subnets are layer 3 concepts. In most cases, there is a one to one correspondence between them -- one IP subnet per VLAN and vice versa.
Typically, you would use a security control that operates at layer 3, such ...
Certainly this is the usual situation for the Cisco and Avaya phones I'm familiar with, as it's usual to keep voice and data traffic separate.
This is from a Cisco SPA504G phone web configuration; these facilities are available through any of the methods for configuring them, through their remote provisioning mechanisms (HTTP/TFTP etc)
You see that the ...
A VoIP phone that chains to another device is a switch, and it negotiates a trunk between the phone and the switch. This happens with CDP or LLDP.
For example, a Cisco switch interface configured as an access interface connecting to a Cisco phone will use CDP to negotiate a trunk from the access interface:
interface GigabitEthernet 1/0/1
switchport mode ...
WS-SVC-CMM-24FXS can be set in the CMM to use a non cisco sip proxy which will allow any PBX capible of working with a sip proxy to manage the calls.
I also looked through the release notes of the 12.2 IOS release and it DOES list the FXS card as compatible.
Mixing versions like this, depending on which version of CME and CUE you are using, will work in some instances.
In your particular case however, it appears that the answer is no.
CME 8.6 will only work with CUE versions 7.3, 7.4, 8.5, and 8.6.
Since you're talking about a plain NM-CUE in a 2811, that module will only support up to CUE version 7.0. Your ...
When you design a wireless solution for VoIP, several things must be taken into consideration:
Jitter: Variations in delay of packet delivery , Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and are usually only effective on delay variations less than 100 ms. Jitter must therefore be minimized.
Latency : Delay ...
In traditional public telephony in USA exists the North American Numbering Plan that defines this:
NPA (Numbering plan Area Code) 3 digits
NXX (Central Office) 3 digits
xxxx (Subscriber Number) 4 digits
If you call whitin the same area code you don't need to dial it, so you can dial NXX-xxxx (7 digits)
If you call to another area code you need to dial it,...
Yes. You can connect 2 IP phones same network (LAN and it should not be exposed to the internet because you'll get ghost calls) and call the other phone by dialing its local IP address, even when the phone does not have a normal phone account.
Here you can see screenshots from such a call. This is a Yealink T28p IP phone not connected to a PBX and no ...