I have 4 remote sites that are using a ASA as thir firewall / router. I'm setting up a full mesh VPN between all the sites. One of the sites have a UC500 and the other sites access that UC over the VPN tunnels. I would like to set up some basic QoS for the VOIP traffic
The site that has the UC will have multiple vpn tunnles coming in from the remote sites. How will I do QoS with voice traffic on that site?
I have a Cisco ASA 5540 running 8.2(5). When I dial a phone on the other of the the VPN the first time I get a blank after it rings(i.e when the voice mail get activated if someone picks the phone up), however works the second and consequent times i dial.
A little background. Two sites A and B connected via IPsec Tunnel. No problems in communication except for the VoIP issue. A Phone in on site A(172.17.168.x) and other on site B(192.168.103.x). Site A and Site B is connected via an IPsec tunnel on the Cisco ASA. First call fails. Second call works. Result of a packet trace is also the same. The UDP packet get drops when tried for the first time but subsequent ones pass.
First time
ASA5520# packet-tracer input inside udp 172.17.168.95 10000 192.168.3.103 10000
Phase: 1
Type: ROUTE-LOOKUP
Subtype: input
Result: ALLOW
Config:
Additional Information:
in 0.0.0.0 0.0.0.0 outside
[code].......
How can I do a VoIP Install/Repair?
View 3 Replies View Relatedhaving some issues. My basic VOIP network I can get to work no problem uner Vlan 1. But when I try tomake multiple basic networks to connect and put them in to diffrent Vlans such as Vlan 2, 3, 4 and conect them the phones now say configuering IP.
View 1 Replies View RelatedWe just purchased cisco 2960 for our VoIP needs and we are using polycom phones, and Phone and Computer will use same port. Since Polycom phones are capable working with CDP protocol and we are hoping to get another switch to expand VoIP network. I found easiest way of setting up each port is as following (from the cisco tutorial)
Switch#configure terminal
Switch(config)#mls qos
Switch(config)#interface fastethernet 0/1
Switch(config-if)#mls qos trust cos
Switch(config-if)#switchport voice vlan dot1p
Switch(config-if)#switchport voice vlan 10
Switch(config-if)#switchport mode access
Switch(config-if)#switchport access vlan 20
Switch(config-if)#exit
My first question,when we are using switchport voice vlan dot1p ,I thought we instruct the switch port to use 802.1P priority tagging for voice traffic and to use the default native VLAN (VLAN 0) to carry all traffic.Do I still need to create a Vlan 20 for data and Vlan 10 for voice ?
Secondly,same tutorial adds these commands as well,Do you think for our set up, using those commands are feasible ?
Switch(config-if)#switchport priority extend trust
Switch(config-if)#priority-queue out
Switch(config-if)#spanning-tree portfast
Switch(config-if)#spanning-tree bpduguard enable
Switch(config-if)#exit
Thirdly,when we get another switch and do the same configuration for the second switch, can I use any port on Switch 1 as uplink without doing any configuration ?
my config and all the show's ive run sofar tryign to figure this out, but the policy map isnt matching the traffic for some reason
View 9 Replies View RelatedI have a little weird request at work. One of our offices would like to split the VOIP traffic. At that office we have a 10MB primary and 3MB backup circuit. Currently the phones are routing over the 10MB circuit. The General Manager would like to use the 3MB backup circuit for VOIP traffic. For the 3MB we have two T1 lines bundled together in a multilink. Configuration is bellow if needed
3MB Circuit
I have a 887 setup as a EasyVPN server, and a 861 as an EasyVPN remote - network extension mode with split tunnelling.This works fine - I can ping and connect to machines across the tunnel.However if I setup a VOIP handset to connect across the tunnel it registers and calls work, but drop after 30secs....I know this is normally a firewall or nat problem, are easyvpns firewalled or natted?
View 9 Replies View RelatedI am fairly new to Cisco, but am trying to configure a 1921 router to give higher priority to SIP/VoIP traffic (Port 5060) than everything else.The connection is only 4Mb and is getting hit hard by video streaming, I don't want to block this, just make a lower priority.Any ideas where I am going wrong?My current config is as below.The IP addresses have been changed for security reasons, but in reality are both in the same range, i.e. are both external IPs, so I am not sure if this is causing the problem. Do I need NAT for QoS to work?
View 6 Replies View RelatedI am tasked to connect my VoIP phones from remote site to my corp site. Basically all remote phones will be registering into a VoIP server in corp site. I have a site to site vpn tunnel established already from remote site to corp site. My hardware includes the following:
-Cisco ASA 5505
-Cisco small business POE switch SF300 24p
-Avaya 2015p VoIP phones
Successfully Register remote VoIP phones to corporate VoIP server 10.30.18.55.I have already configured vlan1 10.30.15.0/24(inside lan) and vlan2 public int(outside Internet) which my dmz only allows 2 per my basic asa licensing.When I connect my phones and register it states "subnet conflict" unable to register.
I am trying to get QoS for my VoIP system setup on several SGE2000p switches and have got a question...How do I define the ACL for RTP? As far as I can tell it will not let me enter a UDP port range for the RTP traffic... And I cant imagine creating rules for each port would be very effective either. So, how can I define an ACL to cover the RTP traffic so I can classify it?
View 4 Replies View RelatedI am encountering delays when making any calls using VOIP. I understand that there is always a delay using VOIP but not as much. I've done a bandwidth test on my network and everything seems to be Ok, we have 4 T1s with 55 Reps using it. There is nothing choking up the network.
View 3 Replies View RelatedTrying to select a router that will work well networking a computer with a nettalk duo, on a limited bandwidth connection: .3/3Mbps up/down. From what little I've been able to find on this, QoS bandwidth control seems critical, yet the list of recommended routers from Nettalk seems to favor routers that don't have this feature. On the other hand, the list apparently hasn't been updated in a year and a half.
View 5 Replies View RelatedRecently at my company we have started using Polycom i335 IP Phones from 8x8 inc, the thing is that while trying to use it directly through a switch it does not register with the server and I get a "url call is disabled" msg on the LCD of the Polycom. On that same switch I am using an Aastra 6755i IP phone, it is working absolutely fine, although it is using the same technology and ports.Now if I connect my laptop via wireless or if I connect a PC through wire and use ICS to share the internet and connect the Polycom to the shared Ethernet port, the Polycom works fine! On the same network Vonage works fine as well. but this Polycom only and only works via ICS?
View 1 Replies View RelatedHow can I connect two laptops via VOIP?
View 1 Replies View RelatedAny VoIP box which has from 3 to 10 phone ports?I'm sourcing the box for a business that wants to connect about 8 phones to the box, then have the calls terminated on our partitioned platform hosted by a third party.
View 3 Replies View RelatedI have a Voip application on my Windows Vista Home Premium. It is called Reliance iCall. I was not able to call from this application for a few months and so i thought there was something wrong with it. But just yesterday i thought of connecting my modem directly to the desktop and the call went through without any problem. It is only when i connect modem to computer through router that this application does not does not do anything.
What could be in this router that is blocking the calls from making a connection? There is also no setting to configure in this VoIP application. I dont know what to do. And also dont want to keep on switching modem's ethernet cable from router to computer and back.
As the thread title says, I just want to gain better understanding by practising the config.
View 1 Replies View RelatedI'm looking for a VoIP solution for one of our remote sites.I'd like to keep it self contained and not attached to our current Cisco Call Manager.I had a couple sites running Asterisk from EvolutionPBX (URL), but it appears they've gone out of business.I've come across many different vendors for a solution, but of course they all claim to be the best. I'm just curious what others are running.What I really liked about Evolution PBX was how easy it was to setup.I basically need about 25 to 30 extensions, after hours voicemail, and a hunt group.
View 5 Replies View RelatedIs it possible to connect an analog phone to an FXS port on a CME router and a VoIP phone to a switch connected to said router and have voice connectivity between the phones? Also, is it possible to connect an FXO port on that same CME to a RJ-11 wall jack to connect to the PSTN and be able to call that VoIP phone as well as the analog phone from my cellphone? I'm trying to tie as I read the CCNA Voice OCG.
View 5 Replies View RelatedI have a cisco RV220W router used in a home environment. Recently I purchased a Siemens Gigaset A 580 IP phone. It works OK so far but I would like to optimize the configuration for VoIP traffic and apply QoS for VoIP on this router. Any guide with the recommended configuration and preferred settings of the same? I have not used QoS previously
View 1 Replies View RelatedI have a Cisco 877 on an ADSL connection. QoS isn't doing the trick -- I need to reserve 200 meg or so of my outbound (upstream) bandwidth for VoIP to end complaints about voice quality. Any example of how to classify SIP, RTP, IAX, and Skype traffic and put a rate limit on anything that doesn't fall into that category? The VoIP phones also are in their own IP range on the LAN side if that would make things easier...or I could even connect them into a specific port on the internal switch in the router.
View 9 Replies View RelatedI want to implement VoIP on my client network, currently the data network is using SG300-28P small business switches for user access. According to my design the IP Phones (Cisco 9971 and 7942G IP Phones) are to connect to the small business access switches while the user PCs connects to the IP Phones.
My concern is I really don't understand how the small busness switches will advertise the voice VLAN to the IP Phones. I understand that the switches are suppouse to use LLDP/CDP for this but it seems the model I have can only do LLDP. The IP Phones and the PCs connected to them will be recieving IP addresses from an uplink L3 managed switch.
how LLDP works (particularly regarding this scenario)? Does it matter if the small business switch is in L2 or L3 mode for the VoIP implementation?
We have to set up voip for our network(for 50 phones not he cisco phones).
I need to just the route the voip traffic to gateway address of telephonic company(1.1.5.7) where they provide us the connectivity for the setination call.
What sort of protocols should i have to enable in pix i saw the concepts like sip, h323, ras, skinny.
We are using only voip for asa and no data or other traffic should be allowed.
inside adrees: 10.10.10.0/24 for all voip phones
outside:121.21.22.1
telephoneic gateway: 1.1.5.7
I am unable to get any configuration parameters to work on my SRP547 for VOIP. ADSL2 works fine. Equipment replaces a ya.com (Arcadyan ARV4518PW) which worked ine albeit with significantly less features. Problems encountered due to the lack of equivalents for some parameters and an overabundance in the new unit which simply don't exist in the Arcadyan product.
View 7 Replies View RelatedI have configured 2 Vlans on the network. 1 - native Untagged for data traffic and 100 Tagged for VoIP.I have 4 SGE2010p switches 2- of them in stack working in L3 mode and 2 connected to the stack via single ethernet links in L2 mode.I have IP phones with trunked ports -Vlan100 tagged and 1 untagged. I have set CoS 5 for Voice Vlan 100 on the phones and CoS 5 to queue 3 mapping (in basic mode) on all switches.But it seems to not work at all.Should I use Advanced QoS mode?Where Can I find some QoS configuration example for Voice Vlan on SGE2010p switches?
View 1 Replies View RelatedI read some papers about QoS. The thing is that I have to implement 30 switches with QoS to connect to a lot of Cisco IP phones. The switches are Cisco 2960S with code image "C2960S Software (C2960S-UNIVERSALK9-M), Version 12.2(55)SE3". Should I use autoQoS feature or manually configure them?. How should I treat uplinks between access switchs and core/distribution? I have a lot of doubts regarding queueing, because all the info I had read its for another model/image.
View 4 Replies View RelatedI am facing a problem with transmitting of VoIP traffic through VPN.
I have a 1921 router in my end where two ISP's terminate and load balancing is done over the ISP'S. I also have a site-to-site IPSEC VPN connection to remote location. Also I am having to analog phones connected to the network through an ATA. My Call manager is in the peer end and has public IP assigned to it. The IP phones get registered when coonected to general inernet connection.
The loadbalancing and VPN is working fine. Now I need to transmit the VoIP traffic over the VPN. I have configured the same but seems not working. [code]
Here is my setup:
Cisco SPA122
Apple Airport Extreme Wireless Router
imac 2.8GHz Core 2 Duo
Thomson DCM475 Cable Modem
The imac is on a desk in the kitchen and is out main computer. The Cable modem, SPA122 and Airport Extreme are in the den just beside the kitchen.
I guess my first question is if the SPA122 has to ABSOLUTELY be attched to the computer? Right now I have it attached:
Cable to cable modem
Cable modem to Airport Extremet
Airport Extreme to SPA122
In this configuration, I can not access 192.168.15.1
im changing the firewall 5510 to 5515, with ASA5510 the incoming and outgoing calls work perfectly, but when i active the 5515 the outgoing calls doesnt work, only the incoming calls work.
As you see on the topology,the flow of calls happens this way:
In the outgoing calls the phone forward the call to the PABX(172.17.3.4), and the PABX forward the call through the ISP LINK to SIP SERVER (10.140.131.208). The incoming calls occur in the reverse path.
ASA 5510 config:
ASA Version 7.0(8)
name 172.17.3.4 PABX
dns-guard
!
!
interface Ethernet0/1
[Code]...
Im changing the firewall 5510 to 5515, with ASA5510 the incoming and outgoing calls work perfectly, but when i active the 5515 the outgoing calls doesnt work, only the incoming calls work.
As you see on the topology,the flow of calls happens this way: In the outgoing calls the phone forward the call to the PABX(172.17.3.4), and the PABX forward the call through the ISP LINK to SIP SERVER (10.140.131.208). The incoming calls occur in the reverse path.
ASA 5510 config:
ASA Version 7.0(8)
name 172.17.3.4 PABX
dns-guard
!
!
interface Ethernet0/1
description ***ISP SIP Network***
[Code]....
I have a remote user who currently has a Cisco VoIP phone that connects into our network via TFTP. He informs me that the phone drops several times throughout the day, with an occasional drop in VPN from time to time. His router has the most current firmware, set QoS rules priortizing the phone and VPN, and decreased the MTU rate, yet he still experiences these drops. He used a VoIP phone from his last job, and never experienced a drop.
A little info of our current setup:
Cisco 2821 Intergrated Services Router, with CUCM version 7.1.3.30000-1
Cisco ASA 5510 software version 8.2(4)
one of my customers wants to implement VoIP in his existing DMVPN Network Topology. I have read about the "Per-Tunnel QoS for DMVPN" but when it comes to configure it on my hub router (Cisco 7206VXR with c7200p-advsecurityk9-mz.124-15.T14.bin) I am lacking the option to set the "ip nhrp map group" command.
My question now is, is it generally not supported by the 7206VXR platform? Or can I get the option by upgrading the IOS to a newer version? If so, which one could I use ?