We help IT Professionals succeed at work.

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

Good day Experts,

Need help with turn server and web rpc.. we are getting "ICE failed, add a TURN server and see about:webrtc for more details" and we trying to get our video chat application to work properly.  At moment it only works if i am on the same ISP with a user..if users are on different isps..the voice and video does not work.  We are running tests using this kink https://www.teledoctor.co.zm/livesmart/client.html

When it works we are on same service provider, but voice video stops on different..we are not experts on this and need help to get work across service providers for our application.  Is there a site where one can test a TURN/STUN server setup?

Kind regards
I am being bombarded with robocallers on both my present voip  phone service through my local fiber ISP which is Elevate Fiber. I also have a second "landline" service through Magicjack. Both are working, but both are victims of incessant robocalls. I have a cell number that WAS experiencing the same difficulties. I signed up for NOMOROBO.COM service for my cell. I have not received a single robocall in four days! Amazing! I wanted to avail myself of the same service for my two voip numbers but both Magicjack and Elevate Fiber do not support that app. How might I go about finding another voip provider?
I have a customer who's phone are IP phones. You pass through them to get to the network. The other day the phones were having problems and the phone company blamed it on a computer. They said that one of the computers was doing an arp call with the destination IP of and it was causing the phone to drop calls and/or reboot.

   I think they are full of it but want to know. Is there a program available that will monitor arp calls that I can continuously run on either a 2012 Server or a Windows 10 client that will show any arp calls going on?
Hello I'm using CUCM 11.5 and using SIP phones. I have my DID's and looks like I have what I need to have setup so that I can connect to my ISTP. but when they dial in, they cannot establish a connection to download their config files.

I wanted to know what else do I need to setup in CUCM? For SIP phones?

We have CUCM V12.5 installed. I have users in London, Paris and Singapore and I am having issues with Call Forwarding.

I am based in London and if I put my 8851 on call forward to my mobile anyone in the London office can call my number and it will forward to my mobile fine.

If someone in my Paris or Singapore offices calls my number they get "Fast Busy". If I turn call forwarding off they can call me fine.

If I forward my number to another internal number it also works fine. The issue just seems to be forwarding to external numbers.

Each location uses its own Device Pools, Partitions and CSS's etc...

My guess is the call is coming to london and then trying to break out of the London Voice Gateway but the format is incorrect. The London Voice Gateway is a Cisco ISR4321 andis attached to a SIP line.

Thanks in advance
Not sure if I have setup QoS for VOIP correctly to interact with RingCentral on my switch.

I know almost nothing about this aspect.  Switch is Aruba 2540.

RingCentral is seeing jitter when they diagnosed it, I want to make sure that the switch is prioritizing the voip traffic at the best priority over data but I have no idea if this is the right way to do it, or if there is a best-practice type of approach?  The phones are on vlan 10, phone plugs into switch, pc plugs into phone, PoE is used.  In this instance I just want to confirm the switch is properly configured to prioritize traffic, if the problem is upstream, it's upstream, I am just concerned about the switch at this point.

I don't know what the dscp mapping table looks like so maybe ultimately I'm missing the priority?

hostname "zzz"
module 1 type jl356a
qos type-of-service diff-services
ip default-gateway 192.168.xx.xx
ip route 192.168.xx.xx
snmp-server community "public" unrestricted
snmp-server contact "zzzzz" location "zzzzzzz"
vlan 1
   name "PCS"
   untagged 1-28
   ip address 192.168.xx.xx
vlan 10
   name "PHONES"
   tagged 1-24
   ip address 172.168.xx.xx
   ip helper-address 192.168.xx.xxx
   qos dscp 101110
no tftp server
no autorun
no dhcp config-file-update
no dhcp image-file-update
no dhcp tr69-acs-url
password manager
password operator
I wonder if you could help us with the phone configuration in Dubai.

We have a 3CX phone system in the Uk
We’ve just rented some space in a service office in Dubai
And have arranged  for Port 5060 (inbound, UDP) for SIP communications and
Port 9000-10999 (inbound, UDP) for RTP (Audio) communications to be opened on their firewall and linked to the IP addresses of the phones in the office.

We are told that the public ip is which can be used to access the services (I don’t under this part)
However the phone are showing a SIP error and not currently working.

We’ve only two full days left in the Dubai office and I am reaching the edge of my understanding.

Would you be able to assist ?

Kind Regards

Get Outlook for iOS
I have been trying to get a Grandstream HT814 to communicate with Sonetel. It works fine when connected directly to the DSL line, but as soon as I put it behind the firewall it stops. It cannot make calls, when receiving a call it will ring, but with no sound.

I have tried with the SIP inspection on and off (in the config below it is disabled)

Cisco Config:

Hardware:   ASA5506, 4096 MB RAM, CPU Atom C2000 series 1250 MHz, 1 CPU (4 cores)
ASA Version 9.8(4)12

no mac-address auto

interface GigabitEthernet1/1
 nameif outside
 security-level 0
 ip address dhcp setroute
 no pim
 no igmp
interface GigabitEthernet1/2
 nameif inside
 security-level 100
 ip address xxxxx
 no pim
 no igmp
dns domain-lookup inside
dns server-group DefaultDNS
 domain-name xxxxxxx
same-security-traffic permit inter-interface
same-security-traffic permit intra-interface

access-list outside_access_in extended permit ip any host xxx
access-list inside_access_in extended permit ip any any

access-list global_mpc extended permit ip host xxx any inactive

pager lines 24
logging enable
logging asdm debugging

mtu outside 1500
mtu inside 1500
mtu Proxy 1500

arp timeout 14400
no arp permit-nonconnected
arp rate-limit 16384

nat (any,outside) source dynamic any interface

timeout xlate 3:00:00
timeout pat-xlate 0:00:30
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 sctp 0:02:00 icmp 0:00:02
timeout …
I need to report the number of call from our cisco Call manager that tells me the number of calls for last year that were incoming between 8am and 5pm and were contact type 1, contact disposition type 2,

I tried doing this from call reporting but it will only report on 7 days at a time?
Looking to see if anyone can assist or has thoughts on Microsoft Teams Voice using AudioCodes M800B to push our numbers coming from our CentryLink PRI to teams. The issue I am having is Degraded Audio on the recipients side, sporadically there will be points in a call where the recipient looses audio for a second or 2. CenturyLink is only a best effort network but our 500mb pipe isn't even hitting about half that and at max we maybe have 10 users on the phone at once. When I look at the calls in teams a large majority looks like they are good but there are about 5% of calls the come back with a 408 and 504 Response code.

I have made sure on my Meraki network I have set up all QOS and I have tried to trouble shoot with AudioCodes, Microsoft, and CenturyLink but so far no luck... my company has invested a large amount of time and money and I have about exhausted all my resources trying to track this down... any thoughts that may help me resolve this would be great;ly appreciated.
I'm setting up a 3CX PBX on Amazon Lightsail, and I'm having trouble with setting up conference calling that will allow external participants to dial in (like FreeConferenceCall.com, or other similar services).

I have my inbound and outbound calls working, so I assume my basic setup is okay. I have one number I purchased from Skyetel (the VOIP provider I'm using), and I have another number that's being ported in from Skype (not yet active on Skyetel).

However, I'm not sure how to properly setup conference calling on 3CX. I have a single extension (00), and I have all of my trunks (inbound and outbound) set with the "Trunk" value of *1949 (the last 4 digits of the number I purchased from Skyetel).

In Settings >> General >> Conference, I set my Conference Extension to 00 (the only one I have), and I've set my External Number to the number I purchased from Skyetel (the one ending in 1949).

Are there other settings I need to host conference calls?
Now that Skype no longer allows a customized voicemail greeting, I'm looking for an alternative.

I have a Skype Number that I'll port over to the new system, so it needs to accept incoming calls on that number. Other than that all I really care about is voicemail. Skype cost me about $100 a year for the number + my plan, and I'd sure like to stick around that price range.

Any ideas?
Hi Experts

I am on a personal mission to under a little bit more about VoIP.

So we run a Gamma \ Horizon system at work. All configured and working via Cisco.

Our Wifi is controlled via Unfi USG, Switches etc. I've plugged in (With Consent) one of our spare Voip phones into a spare port on one of the Unifi switches and configure on it own Vlan etc.

The phone can call external numbers no problem, but if I try to ring an internal ext the internal phone rings but once connected I can not hear the person on the other end.

I get that something is blocking it or its not configure I just do know where to start trouble shooting,

Maybe someone could point me in the right direction?

Please let me know, the best USB Headset for Skype, MS Teams and Cisco WebEx calls, under $100.
Also, share any Gadget for USB to 3.5mm converter.
I need to re-IP all the voice VLANs in a company. My first thought was to just renumber the SVIs at each site related to VOIP. But that would then cut me off from being able to reset the phones from the Call Manager because the gateway no longer works. OR would it be the case that losing connectivity to the Call Manager, the phones might just reboot themselves?

If not that - might there be a way to recycle the inline power at the switches to force the phones to reboot? e.g

int range Gi 2/0/1 - 47
    power inline never


    power inline auto

Any other thought on the most efficient means to reboot all the phones on a switch when they can't talk to the Call Manager?
Need Speech to Text service

I have an iPhone 6, a MacBook and a Windows PC. I need to find a service which lets me speak into some device and get the text version emailed to me. I do not have time for a service, which can cost $1/minute and take 12-24 hours. I need it immediately, which means the text creation needs to be near real time.

I will pay for this service, if needed. I think $1/minute is the highest I could pay.

by "some device" I meant:
1) My iPhone 6
2) A microphone hooked into a Mac
2) A microphone hooked into Windows

Need a quick web UI for a great speech recognition API

I sampled an AWS back-end API (using their demo) and found the quality to be excellent. Meanwhile, I have heard Google and Watson also have great API's.

But, my friend, who can no longer type into a keyboard, can not find a way to access any of these great API's.

Can you tell me the names of these services?

Do you know of any consumer focused front ends that would provide access to these awesome API's?

If I decide to throw together a quick front-end, which API is easiest to develop with?

I have a Cisco Voice Gateway 4331 that handles all of our calls in conjunction with Cisco Call Manager.  The voice gateway has a PRI circuit connected to a port and three POTS lines using the remaining three ports.  In this example, I want to have the internal extension 3337 use the specific port of a POTS line on port 0/2/2.  This is for a fax machine (attached to CUCM via an ATA 190) that only sends and I am having trouble with it being reliable over the PRI.  I was hoping to tie it to a POTS line to avoid trouble.

I tried the following and it did not seem to work correctly.  I feel like I am missing an important component:

dial-peer voice 3199 pots
desc ***** Send faxes over pots lines for mailroom *****
preference 1
answer-address 3337
port 0/2/2
forward-digits all

In the above, I am attempting to identify the internal extension of the fax machine (3337) so that it can be directed to use the POTS line on port 0/2/2.  Is there another set of commands that I might be missing?

I have Nec ITX-3370-1W(BK)TEL POE phone and when I attached it to Cisco 3750 POE switch the phone not working while when I attached Nortel 1120E or 1140E they are working.

Any idea to fix the problem?

Good Day,

I recently got a E1 line through digium gateway G100, then bought 3 digium d60 and 1 d80 ip phones, and then a conference phone for my office.

I have been using Wifi for over 1year now with zero issues. Use ubiquiti AC Pro, 2 on the floor. open space of 289sqm.

How can I deploy the IP phones without without having to run LAN please?
I'm dealing with performance issues with a VOIP phone system.
The VOIP service provider provides a dedicated internet connection for VOIP and the "PBX" is externally provided by the VSP.
Since the PBX is external, even internal extension-to-extension VOIP calls cause external traffic.
We provide a dedicated VOIP firewall in the form of an RV320 followed by cascaded SG300 switches - configured with a VOIP VLAN with QoS set up.
We provide a dedicated internet connection and firewall for site data - independent of VOIP traffic.
There are 3 sites, each one with separate internet connections, firewalls, etc.
The largest site has about 20 phones and 25 workstations.
The smallest site has about 6 phones and 10 workstations
The middle site has about 9 phones10 workstations.
Data traffic is modest.

I believe the VOIP system is working overall as intended so the "problems" are a matter of service quality I'd say.
Problems are intermittent and include:
- audio is heard at one end and not the other.
- a very loud "screeching noise" is heard at one end or the other and can be audible at one or both ends.  This is reported to be rather high-pitched and not like loud TV white noise.
- some incoming calls don't arrive on site and go directly to voice mail.
Overall, it's reasonable to say: "while the system seems to "work", service is unacceptable".

Since the 3 sites are each independent of the other re: VOIP, if all sites behave similarly (re: problems) then one might …
For years I have used Plantronics Supra binaural headsets along with the matching Plantronics headset amplifier/interface M10 or MX10. ( I am not at that location now.) I have the requirement that my phone audio be crystal clear at all times. I have never had any problem with clarity until yesterday. I never had considered VOIP because my internet speed was not ideal. Several months ago I got fiber and my up and down speed is 1 GIG with pings at 2ms. With this super speed I thought that VOIP would be an acceptable choice since it would save me more than 50% of my phone bill. Now that it is installed I am told that my transmitted voice is somewhat distorted and there is some sort of slight crackling in the background. I cannot live with this problem. I spoke to level one of tech support last night and he confirmed that I was indeed distorted. I have another line that utilizes the MagicJack. I phoned the tech on that line and the distortion was still present. I also switched from my headset and amp combo to a regular phone on the business phone system (Avaya Partner) and the distortion is still there. Level two is supposed to get back to me today and begin troubleshooting the problem. I was just wondering if any Expert has encountered this difficulty before? With such incredibly high isp speed the is the last thing that I had expected. If an Expert has any ideas please let me know.  

Configuration wise: On the isp's router there are two phone jacks. I go from jack One …
The phone is Polycom VVX 350, provisioned by RingCentral.

Is it possible to somehow program the Polycom phone to produce a distinctive ring if a specific number calls.  A mobile phone can do this.  I wondered if there is any way that I can get this feature? If not Polycom, is there another brand of VOIP desk phone that has this ability?

I am replacing a Cisco 2960 switch with a 9200. After copying & pasting the run config from the 2960 the interfaces are missing the following commands;

srr-queue bandwidth share 1 30 35 5
priority-queue out
mls qos trust device cisco-phone
mls qos trust cos

I also see that in our AUTOQOS policy-map for our Cisco phones that the 'police...….' configuration line detail has not pasted across

I am wondering if these commands are handled differently in the 9200 and have to be reconfigured accordingly.

Expert advice on this would be greatly appreciated.
Good day. I have a Data and Voice VLAN, with the PC's getting DHCP from Microsoft DHCP Server (2012) and the phones from a Linux DHCP. I want to decommission the Linux DHCP and move everything to Windows DHCP. Phones are Polycom Soundpoint IP 331, Server is 2012, Voice VLAN ID:2. Please assist to configure phones to get DHCP from Windows server.

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

Top Experts In
Voice Over IP