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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).

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I've been struggling with VOIP call quality. One thing I notice is where calls to PSTN or Conference are made from work on the west coast or from home on the west coast - the traffic takes a 100ms trip to the east coast to get to the Skype voice control and RTP gateways. Is there any way this can be altered so you can use gateways on the west coast to reduce hops and delay to the Skype voice gateways?
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Hello

We are using AVAYA IP Phones.

When I open AVAYA IP Office Manager, I get the following msg:

"The security certificate will be expired in xxx days."

I have found the following solution on internet:

Security settings > system > identity certificate > delete the certificate and save security configuration IPO will generate a new self signed certificate.

Actually, I am not an expert in managing IP phones and have limited knowledge regarding this, so I am a bit confused and afraid of applying the above mentioned solution, because after applying if any other problem appears then it might be a problem for me to solve, then I have to again go through internet finding the solution of that newly created problem.

Any advice and assistance shall highly be appreciated.

Thanks
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Have been on a Cisco/Linksys E3000 router for a few years.  My Mitel 5360 IP phone has worked flawlessly connecting to my office the entire time.

Replaced my router with a Netgear Nighthawk X4 R7500V2.  The IP phone gets an IP address (DHCP) but hangs at "Contacting Server."  Cannot connect to my office.

Reconnected the old router and the IP phone works fine so it's not the phone.  

Suspect I have to open up a port on the new router but have no idea where to start.

Any suggestions appreciated.
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Normally I've seen MOS scores on a 1 to 5 point scale. But in Skype for Business Call Analytics they're expressed a decimal. e.g.

Average network degradation 0.009717 MOS

What are they saying? Is that a 4.93 MOS score?
ScrnGrab2671-171018-17.21.jpg
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I am curious if there may be a better phone system backup strategy other than what I am currently doing. My current backup involves  rerouting the main number to a series of cell phones that are configured in a chain (no answer/busy forwarding enabled on each)

I have noticed there are some online phone system backup solutions. In the event that our onsite phone system goes offline for whatever reason, I need a solution that is cloud based, and does not require any local hardware other than a computer with internet access. It would need to be able to accept at a minimum two telephone numbers that would be forwarded. Does anyone know of any service that could handle this?

Thanks in advance!
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Hi All,
 
We currently have Business Voice Edge VIOP from Comcast which is their proprietary voice platform. They have provisioned and require a 50Mb circuit over their fiber backbone to our office to services SLA for their voice platform. Thus far, as far as reliability, I have to say that we have had no real issues with call quality over the past year of usage.
 
Management has decided to move offices earlier than expected, and we overlooked Comcasts terms and conditions regarding portability of service to locations that do not currently have a Comcast fiber  backbone in their building – which the location we are moving to does not have Comcast fiber. They are also not willing to work with hhus to temporarily provision over another circuit. At this point, we have three options – ranked in order of preference, and I wanted to know if anyone has experience and any recommendations to help in making the right decision. Here are the scenarios:
 
Upgrade with Comcast to the new location and wait 6 months for them to build out their own fiber (includes city permits) to the new office.
One of two options in this cast to get service to our new office:
                                                               i.      Implement RingCentral month-to-month as a temporary VOIP platform while we wait. Forwarding temporary numbers to main numbers.
                                                             ii.      Implement a Ethernet Dedicated E-Line (point to point) between our …
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Hi All,

We currently have Business Voice Edge VIOP from Comcast which is their proprietary voice platform. They have provisioned and require a 50Mb circuit over their fiber backbone to our office to services SLA for their voice platform. Thus far, as far as reliability, I have to say that we have had no real issues with call quality over the past year of usage.

Management has decided to move offices earlier than expected, and we overlooked Comcasts terms and conditions regarding portability of service to locations that do not currently have a Comcast fiber backbone in their building – which the location we are moving to does not have Comcast fiber. They are also not willing to work with us to temporarily provision over another circuit. At this point, we have three options – ranked in order of preference, and I wanted to know if anyone has experience and any recommendations to help in making the right decision. Here are the scenarios:

1.)      Upgrade with Comcast to the new location and wait 6 months for them to build out their own fiber (includes city permits) to the new office.
a.      One of two options in this cast to get service to our new office:
i.      Implement RingCentral month-to-month as a temporary VOIP platform while we wait. Forwarding temporary numbers to main numbers.
ii.      Implement a Ethernet Dedicated E-Line (point to point) between our office via layer 2 routing.
2.)      Terminate with Comcast and incur termination charges – something that we don’t want to do.…
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We're running into call quality problems and I think part of the problem is that the VOIP real time packets are traversing many hops to the other side of the country before they finally make it into Microsoft/Skype network. My question: Is there any way to choose which gateways to use for peering with Skype for Business 365?
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Hi ALL,

We're trying to plug in a phone into a switch that is primarily used for data but was configured for data and phone, however, when the phone is plugged in it does connect to the network. Below is config from the data switch configured for data and phone and a switch configured for only phones ( or at least that what we use it for).  Please let me know. Thank You.

Switch configured for use with phones - both data and phone work

interface FastEthernet2/0/39
 switchport access vlan 10
 switchport voice vlan 20
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast

Switch configured for DATA - Only Data works (what do we have to configure for phones to work?)


interface GigabitEthernet3/0/46
 switchport access vlan 10
 switchport voice vlan 20
 spanning-tree portfast
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For Skype for Business O365 MS offers a Call Quality Dashboard that shows quality trends. e.g. 1000 good calls, 20 unclassified, five poor calls. But I'm not seeing a way to search what were those five poor calls and when did they happen, what was the latency or jitter, etc etc. Am I missing something? How do I drill into call quality with this tool? Or are there other tools that would get this done?
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We just connected a new Cisco SG500-52MP to an existing Cisco SG500-52MP.  We use both VLAN 1 as the default and VLAN 20 for voice.  Via the web interface, on the new switch, we tagged ports 1-48 for VLAN 20.  We connected the new switch port 48 to the existing switch port 1.

When we connect a VoIP phone on the new switch, the phones do not connect.  If we connect the phone to port 1 on the existing switch the phones work so I am sure we are missing a step on the new switch.  What are we missing?
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I hear that waiting until the URL for a meeting arrives for first set up WebEx can be time consuming. So, is there a way to do this initialization beforehand?

Thanks.
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We have an HP Procurve 2920 POE switch. We have been having issues with our new VOIP service so I am becoming very familiar with our switch configuration. One thing I am noticing is that while POE seems to be working to all of our devices, only 3-4 of them show port status "delivering" in the switch. The rest of the devices show "searching". Any ideas?
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Small network with no vlans - starting to have voice quality issues on Mitel phone system (5000).  Two Cisco SG300 switches and a Catalyst 2960.   I'm thinking splitting the voip traffic out onto a VLAN but need to know where to start as this is new territory for me.  

What needs configuring on the switches and what needs configuring on the phone system?
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I am attempting to force all ShoreTel IP Phones onto the voice VLAN.  However despite various attempts the connected devices remain on the default VLAN.  Due to computers connected to the pass-through Ethernet ports on the phones we cannot use the primary VLAN for the port.

I have a phone with a computer in pass-through connected to gi6.  I have been attempting to coax the switch to place the phone on VLAN 10.  Yet I consistently receive the following output:
switchb#sh mac add int ge6
Flags: I - Internal usage VLAN
Aging time is 300 sec

    Vlan          Mac Address         Port       Type
------------ --------------------- ---------- ----------
     1         00:10:49:45:8c:26      gi6      dynamic
     1         08:2e:5f:07:b1:7d      gi6      dynamic

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Running Config:
switchb#sh run
config-file-header
switchb
v1.4.8.6 / R800_NIK_1_4_202_008
CLI v1.0
set system mode switch

file SSD indicator encrypted
@
ssd-control-start
ssd config
ssd file passphrase control unrestricted
no ssd file integrity control
ssd-control-end 
!
vlan database
vlan 2,10,65,200-201
exit
voice vlan id 10
voice vlan state oui-enabled
voice vlan oui-table add 0001e3 Siemens_AG_phone________
voice vlan oui-table add 00036b Cisco_phone_____________
voice vlan oui-table add 00096e Avaya___________________
voice vlan oui-table add 000fe2 H3C_Aolynk______________
voice vlan oui-table add 001049 ShorTel
voice vlan oui-table add 0060b9 Philips_and_NEC_AG_phone
voice vlan oui-table 

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got a missed calls from last two months, This evening I got a ring from +381 628302791 at 4:08.Whoever calling rings 1-2 times and disconnects immediately.
searched on google Which country code +381, It's from Serbia I shocked why I got calls from this area.
I would like to know, is this going to be a problem? Like hacking etc??
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I'm setting up an application that integrates with asterisk. From my webapp, a user registers and connects to the asterisk. I have successfully setup the billing of user A calling user B with the a2billing. Now there is another feature of my app that requires B-Party(the callee) to be billed for receiving calls. I've been searching online for any clue but couldn't find any.
Please can someone with the a2billing knowledge help out with how to get this done?
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I have a subnet (10.201.x.x) used for VOIP. The .1 gateway resides on an HP 5400 series switch in order to provide multicast services to a number of internal networks across our WAN. We have just installed a Cisco Express Gateway for Jabber clients to utilize VOIP services from home, across the Internet. I set up a PBR on the voice vlan to use 10.201.0.3 for a gateway when calls come in across the Express Gateway and to use 10.201.0.1 for a gateway when communicating with our internal networks.

A key server with IP 10.201.0.5 has Trusts established for LDAP authentication to our Call managers. When the PBR is in place, the Trusts do not work for some internal domains because the .5 server is now routing to the Internet for DNS rather than looking to the local conditional forwarders to identify the domain controller of the internal network attempting to authenticate.

I know you can set up a PBR to be port specific but the Jabber client and Expressway use many tcp and udp ports to communicate. Is there a simple way to force the .5 server to resolve all DNS internally yet route all other traffic pertaining to a Jabber connection out the 10.201.0.3 gateway?

Here is a snippet of what is not working:

Policy pbe 10201-routing
     150 class ipv4 "nls-1916814"
      action ip next-hop 10.200.200.100
      exit
     160 class ipv4 "kms-172"
      action ip next-hop 10.200.200.100
      exit
     170 class ipv4 "10201-Internet"
      action ip next-hop 10.201.0.3
   

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I have a client with about 30 users that work at home about 50% of the time.  They continue to have connection issues logging into Citrix, RDS and now a Parallels server.  The problem is not what we are logging in to, which is what they think.  We have the users mostly hardwired to their router so we aren't dealing with wi-fi issues.  There are various internet vendors (Comcast, Verizon, ect) mostly in the DC area but some scattered across the eastern states.  Everyone works fine for awhile (like the last two weeks, barely a problem) but then we have various issues that turn out to be a user working from home, locking a file or getting disconnected repeatedly.  We might have 10 users logging in from DC office to Berkeley without any service disruptions during that same time.  

Before we start ripping servers out, I think we really have to make sure that the WAN is working correctly.  That starts with home internet.   Most of them are running a company provided VOIP phone at home.  It doesn't seem related but maybe it is.
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Hi,

We have been having audio trouble with S f B Online phones that are on the usual SonicWALL TZ600 WAN>LAN.  Even though VOIP settings, Qos and bandwidth management are enabled.  We still get delay, echo and other audio un-pleasantries.  I decided to move the phones, Polycomm and an Audiocodes to a DMZ port on the TZ600.  They are now sounding better but I am concerned about the security.  It seems that Polycomm has updated firmware that locks off port 80 and 43, but I am wondering about other possible danger.

Computers remain on the filtered and trusted LAN.

This gives up the "Better together over Ethernet" stuff that happens if you connect the computer through the phone switch, but the audio suffers and that is not ok.  We are used to excellent PRI audio and will not suffer worse.

Other environmental issues are that there is also a site-to-site VPN between another TZ600 at the main office (this is a branch office) and I believe is Comcast 150/10 cable modem service here and 100/10 at the main.  Another reason I moved the phones to the DMZ was to give them DNS Servers off of the VPN, and their own DHCP range, since the VPN needs to have the main office dns server for connectivity.  So WAN IP is fixed from Comcast, LAN is 192.168.168_ and DMZ is 10.101.101_

I am convinced it is not bandwidth, but deep packet inspection and dns that is primarily at fault with the audio.

Points given for most helpful response, thank you.
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I have a Cisco WS-C4503 chassie with two WSX-4148-RJ45V 48 ports blades.  When I connect one of my NEC VOIP phones up to the system I do not get any power to the phone from the port.  I have check to ensure port POE is set to auto.  Here are some of my configs, I assume these blades are POE since they have DC In-line on them.
_________________________________________________________________________
Cisco_MFG#sh environment status
Power                                             Fan      Inline
Supply  Model No          Type       Status       Sensor   Status
------  ----------------  ---------  -----------  -------  -------
PS1     PWR-C45-2800AC    AC 2800W   good         good     good
PS2     none              --         --           --       --

Power supplies needed by system    : 1
Power supplies currently available : 1

Chassis Type : WS-C4503

Power consumed by backplane : 0 Watts

Supervisor Led Color : Green

Module  1 Status Led Color  : Green
Module  2 Status Led Color  : Green           PoE Led Color : Green
Module  3 Status Led Color  : Green           PoE Led Color : Green

Fantray : good

Power consumed by Fantray : 30 Watts
_________________________________________________________________________

Cisco_MFG#sh power inlin
Available:1400(w)  Used:0(w)  Remaining:1400(w)

Interface Admin  Oper            Power(Watts)     Device              Class
                            From PS    To Device
--------- ------ ---------- …
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We have an issue that has dragged on for a few weeks since our migration from internal PBX to Vonage VOIP. We have dropped calls...inbound calls go straight to voicemail....inbound calls blink ( quick 1/2ring on phone) and then go straight to voicemail.....this is intermittent...We have some inbound/outbound calls that work just fine but everyone is having the issues daily. We have been working with our firewall support team and Vonage but no changes so far...

Hardware
Sophos XG210 / HP 2920-48G POE / Comcast Business Class Internet 150/35(normally) / Polycom VVX411 phones w/30 computers chained from phones to network / 5 servers

Troubleshooting to this point:
 
  • Firewall - IP of all phones in group on firewall that allows all traffic in/out
  • Firewall - prioritize all voip traffic
  • Firewall - tech found some MTU issues, changed MTU speeds on lan port to switch
  • Switch - no changes made but did see that lan port from switch to firewall shows following error :excessive undersized % giant packet

Any assistance that could be given would be great! This is a hot button issue for us!

Vonage Dump 2 Vonage Packet Dumps
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Hi,

We have VOIP service through intermedia, they do not offer call recording. Is there some third party service i can use to record calls?
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I see sub-domains available. What's it do?
0
What is the best practice of backing up Cisco Call Manager 11.5 ? (VM)
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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).