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).

Share tech news, updates, or what's on your mind.

Sign up to Post

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 …
Bootstrap 4: Exploring New Features
LVL 13
Bootstrap 4: Exploring New Features

Learn how to use and navigate the new features included in Bootstrap 4, the most popular HTML, CSS, and JavaScript framework for developing responsive, mobile-first websites.

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.
In doing business networks, I've encountered quite a few VOIP installations.  
Many of them are provided by the ISP provider and run on the same office subnet as the computers.  Even these have troubles from time to time but are really quite simple.
Others have the VOIP set up with a separate internet connection solely for VOIP, provided by a separate VOIP provider, and are isolated from the office network with a VLAN.
In the latter case, responsibilities get blurred.
So, I'm wondering if there is a VOIP industry best practice?

Here's an example:  
One desk's phone often has a screeching noise "like grinding metal" in the audio that is audible on both ends.
The phone has been replaced a couple of times to no avail.
Other phone on the same switch end has no trouble.
I've started a Wireshark capture of the traffic.

Another example:
Calls are sometimes such that audio is only working in one direction.  So, only one participant can hear anything.

We know and understand the network.
The VOIP provider understands VOIP and their system.

What is common and best practice for dealing with things like this?
How much support and service should we expect from the VOIP provider?
Who leads in the investigation?

I'm not interested in finger-pointing, just trying to calibrate expectations.
I can well imagine that this is contract-dependent but this must be a common situation with a wealth of experiences.
I am using Freepbx 14 and working fine but I got thousands of attacks and in Intrusion Detection, my public ip  has been blocked sometimes and because of this calls are not working. I am using fortigate firewall and opened the 5060 to 20000 ports for the FreePBX so My question is 1. are ports forward mandatory for inbound route ( if I change the sip registration port from 5060 to other and do same with the trunk provider ) . Please let me know how I can make this FreePBX more secure so call disturbance would not occurred in future.
AWS, Speech Recognition and AWS

I have heard AWS has an excellent Speech Recognition web API, and wonder if anyone knows of a complete speech recognition app which uses this AWS API?

Apparently, the demo emails you the translation and I hear the translation is great.

Anybody got any leans on an app that can handle speech to text?

Good afternoon everyone.   I was wondering if anyone had any experience using the My Fax application.   Currently one of our customer is using the service to send outbound faxes directly from their EMR application.   The majority of the fax numbers associated with the EMR application they send to are coming through without a problem.   There is one fax number (which I believe to be an analog line) when sending , the EMR application shows the fax sends and completes the send however only partial pages are coming through.   This is only happening with 1 fax line out of the rest.   According to the vendor they did state this.

"Good afternoon. Pls note that this change reflects ALL faxes sent. In our experience, this raises a few additional issues:
1) Additional Support Calls in to our team as to why the failed confirmations are delayed (ie could take up to 30 minutes to produce a failed confirmation)
2) If the line we are dialing is of poor quality and cuts off multiple times mid transmission, we will still try multiple times to deliver that fax meaning that they would receive partial faxes…."

Where specifically can I look to find out where the break is happening?   Is there a way to verify if the line quality is poor?  Or if not, and the fax is making it to the destination could the drop happen on the way they are processing the inbound transmission on their end?

Here is some additional information from the vendor..

I have investigated and tested the number …
Hi All,

We have an older Mitel 3300 using predominantly Mitel 5212 phones. We have a dedicated VLAN for voice traffic, which is tagged on our switches alongside whichever VLAN is providing the network for the computers. We are having a strange problem in which when a phone call is made, one side is unable to hear the other (in reverse it is fine). After a few seconds this will sometimes resolve itself, or they will need to hang up and try again. On the retry both sides can usually have full conversation as normal. This happens randomly throughout the day to different number / handsets that are plugged into various switches throughout the organisation. We have had a look at the switches and they don't have any obvious errors / are not reporting they have maxed out on throughput.

We have had our support engineers look at the phone system and they cannot spot anything obvious with regards to the fault.

Any clues as to where to start diagnosing this.

Dear Experts,

We have two Fanvil C600 phones in the office.

One is working well, the other is just showing a blank screen with the led indicator "red".

I have done firmware upgrade (pressing back and home screen button) and reboot the phone but still showing black screen.

I am not sure of wiping out the config as i have not done any backup.

Connected phone to network, the wan ports says "1000 mbps" but I am not able to ping the static ip address

The phone displayed an IP, but I cannot ping it.

Connected to lan port of phone but there is no connection.
The user lives in an apartment complex that provides phone service as part of the rent. They were able to connect their Panasonic wireless home phone with 3 stations to this system.

However, even though the wireless phone receives calls, the caller ID isn't working.  Most inbound calls show the ID as DID/DOD.  Outbound calls show the town or just the phone number.  I called my iPhone, and my iPhone recognized the number as being on my contact list, and thus showed the name.

In addition, the answering machine built into the main station usually doesn't pick up.  Management said that an answering machine should work.  The Panasonic HAS an answering machine.

Is there anything we can do to get the caller ID and answering machine portion of the Panasonic phone working?

PMI ACP® Project Management
LVL 13
PMI ACP® Project Management

Prepare for the PMI Agile Certified Practitioner (PMI-ACP)® exam, which formally recognizes your knowledge of agile principles and your skill with agile techniques.

Teams VOIP not ringing on all devices.

I am currently doing a POC to test Teams VOIP and after I moved my user to teams only mode I got the calls to route thru my teams desktop app on my laptop but I am also logged into teams on a Yealink Teams Physical Phone and the Team Mobil App on my iPhone. Is there a way to get all 3 devices to ring? Or how to pick what device will ring?
We have a Fortinet FortiWifi 30e connected to BT fibre.  There is a single software switch including all 4 ports. There is a DHCP server and scope (192.168.5.x) running on the internal interface.

I have defined VLAN_10, set the VLAN ID as 10 and created a second DHCP server and scope (192.168.10.x).

There is a single cable running from port 1 on the Fortinet to a ubiquiti 16 port POE switch.  5 yealink T46s phones are directly connected to the Ubiquiti switch and 5 laptops connected via the second port on the back of each phone.  All devices get IP addresses from the internal range i.e. 5.x

When I set the VLAN tag on a laptop to 10 it will get an IP address from the 2nd scope i.e. 10.x as expected.

BT provided the phones and will not allow us access to the configuration so I can set the VLAN tag on the phones to 10.  They have suggested that we enable LLDP-MED for VLAN_10 and this will force the phones to pickup  an address from the 2nd scope.

Any suggestions as to how this is done? (preferably in noob speak)
Hi there,

I just upgraded the firmware on one of our PowerConnect 6248P VOIP switches to version  But even though the switch is up and running, I can't connect to the management interface website anymore.  Every time I go to the switches management IP from a browser, the site never comes up.  Is there a quick solution to getting the site back up again?
Say, the following device :  D-Link 3G FLLA Wi-Fi Model DWR-720/PW   takes a SIM card and allows one to mkake telephone calls from the built-in handset. It also offers wifi to the user. We are loking for a device with similar functionality as described, except that it does not need a handset but rather a FXO Port allowing one to connect in a regular POTS.
The important features are the Wifi for the end user and the ability to plug in a POTS. Cost is a factor and availability in South Africa is preferable. Kindly suggest devices ofering this functionality.
PS.  This device is being used as a replacemenent to a traditional analog telephone
How do you configure a Polycom VVX 400 to stop switching the time and date on and off?

Visual: https://www.youtube.com/watch?v=X4nQjxz3e28

The Polycom phone  is linked to Skype for Business.
We have a yealink IP SIP phone that needs to connect from the outside.  We have set up the phone and tested it internally and it's good so we moved it externally.  

In the phone, we edited the account and put in the public IP of our router (a Sonicwall NSA4600) and waited for the phone to register.  It fails to register.

I've checked the ports and I DO have the right ports open on the Sonicwall but I've also read a lot of posts about people having trouble with SW and SIP phones.  So, after reading, I have made the changes suggested on the posts I've read and still no joy.  

I'm using https://www.yougetsignal.com/ to test 5060 and it reports that it's closed.

Does anyone have any insight into how to make the SW work well with the SIP phone?

PS:  An alternate port scanning tool tells me the port is filtered.  So, I'm looking up how to turn filtering off in the sonicwall for this service.

QoS Expedited Forwarding (EF) value of 46

Can someone explain the Value 46 of EF  ,where did 46 come from ?

reading online I found this :
 "DSCP 46 is backward compatible with an IP Precedence value of 5 as seen in the following binary pattern: 101110 = DSCP 46 "

Not clear enough though

Any QoS  to elaborate on this ?


I am a beginner with 3CX, and unfortunately I'm having issues configuring my new Grandstream GXW4104 gateway. I can get outbound calls to work, but for the life of me I can't figure out how to get the inbound calls to work properly. I've spent 2 days trying different things, tried several configuration guides, read the forums, been all over the internet, but nothing seems to work. I did have 1 successful inbound call routed to an extension (immediately after completing a re-configuration), but was unable to make a 2nd inbound call - it just rings out.

Strangely, after a standard configuration (as per the 3CX guide), with no inbound rules, inbound calls will consistently go to the operator, which is by default sent to voicemail, as I have no phone set up for that extension. If I then create an inbound rule or change any settings, no inbound calls come through. Changing the settings back to how they were also results in no inbound calls! I can only get it to work again by doing the gateway configuration from scratch!

Apart from the situation above, looking at the 3CX activity logs shows no activity when an inbound call is placed (log set to Verbose). Within the GXW4104 web interface, it shows that the line has a call coming through, yet 3CX reports nothing. I have set up the syslog server in the GXW4104, and from my untrained eye, when a call comes in, it is spitting out heaps of data, yet 3CX activity shows nothing. The gateway status always has a green light …
I have a digital land line for my business, and I'd like to add an auto-attendant to that phone line, without changing my hardware, my existing phone number, or my telephone company.  The telephone company can convert my phones to a VOIP system, with an auto-attendant feature, but the cost is prohibitive.  I have heard about using Google Voice, and a VOIP converter device.  Would this be a reasonable solution, or is there another suggestion?
Build an E-Commerce Site with Angular 5
LVL 13
Build an E-Commerce Site with Angular 5

Learn how to build an E-Commerce site with Angular 5, a JavaScript framework used by developers to build web, desktop, and mobile applications.

I'd like to know how to be able to subnet a PepLink Balance one router for less than 50 users. More concretely I'd like to have the IP phones on a  separate subnet. How can I go about that?
Cisco 7941 phone Registration Rejected.

Call Manager is good.
DHCP is good.
Did factory reset.
Tried SCCP, no go.
Went to SIP, no go.

I am attempting to configure VOIP multicast paging for a member school district. Currently, when a phone page is initiated, the green speaker light on the phone comes on and the mic red (mute) light comes on at same time.However, the page is not able to be heard on any phones. I am told this did work at one time but no one seems to be clear when it stopped working. As of yesterday, I upgraded their WAN switch and I am attempting to get the paging to work again.


Cisco Informacast server (Handles paging) is located on the WAN and is directly connected to a  vlan 110 port (near-end)
VOIP phones are located at the district on the far end of our WAN (far-end)

Near end switch connected to Informacast server on vlan 110:
HP 5412R , J9851A running KB.15.17.0007

Far end switch hands all vlan 110 VOIP traffic off to VLAN 72 (Phones):
Aruba 3810M, JL071A running KB.16.07.0003

Vlan 110 is configured on both switches and vlan 72 is only configured on the far end.

Obviously, the config is non-working (copied over from old switch) and has been modified in an attempt to resolve the issue but I will post for reference and advice on what to change.

near end switch vlan 110 config:
vlan 110
name "VLAN110"
untagged A3,B5-B6
ip address
ip igmp
ip igmp forward A1,A3-A24,B1-B2,B4-B8,B10,B12-B22,C1-C24,D1-D24,E1-E24,F1-F22
qos priority 7
forbid B2
What is the best Opensource chat or team servers that you can install on your premises ? Something like Matrix (Synapse) , RocketChat or Zulip?

Something that supports Chat and PBX integration.
I am looking to roll out a FreePBX phone system for one of my clients. I have experience setting up FreePBX itself, though I'm not sure which SIP provider I should go with to host my phone services and numbers.

Do any of you have SIP providers you would recommend, or have any other tips I should keep in mind while rolling out VOIP services for business?

Thank you!
Ive been researching refurbished Cisco Voice switches.  The pricing is compelling given the abilities within the equipment.  They could potentially solve a lot of problems.
The location where they will be used needs to be provided with a lot of uptime.  They are limited with budget.  The Cisco equipment is at or near end of life, but can be warranted still by Cisco.  the model numbers are the 3750X with full POE.  My research indicates they have static routes and more, but some of the queries I have seen shows that people have had problems with the same said static routes.  Does anyone have these, or used these in the past, or present. If so, is there any issue I need to be concerned with regarding Static routing.  I must have static rotes as there are about 8 subnets within the two campus settings that are to be ties together with MPLS.

If anyone has any Idea it would help immensely.  The issues is highly important as I must make the purchase today, but have this question.

Thank you

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).