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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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Does COMCAST have poor quality phone service?

I switched yesterday from Verizon FIOS to COMCAST and had terrible telephone calls today on the land line.

I think we were sold >100 GPS upload speed. So, I doubt the poor quality is due to a slow internet connection. But maybe the old Verizon FIOS phone number was what they called a POTS line, a Plain Old Telephone Service line. Those were the ones where you cold hear pin drop.

Did I move from a POTS Line to an Internet based phone? And is that the cause of the major drop in quality?

My new modem is...

Xfinity
Dual Band Wifi
802.11.ac

The phone cable goes into the back.

Would an Internet phone do better?

Is there anything I can do?

I really despise having poor phone quality and may go back to Verizon to escape a bad connection.

Suggestions?

Thanks
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skype windows10
automatically adjust microphone settings checkbox

does this reduce background noise

does this cause moments of silence

will I do better when speaking to a machine (voice to text)

what else does this checkbox do when checked/unchecked

automatically adjust
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I have a network with Cisco Catalyst 2960X switches. We are rolling out a new phone system in phases. We want to keep the old system in place as we put in the new phones. To this end, we need multiple voice VLAN's. Another question on here pointed to a solution using MAC Authentication Bypass, but it did not give an example configuration. I am not familiar enough with the VoIP side to configure this, can someone please assist? (BTW, the other phone system is Allworx.)

https://www.experts-exchange.com/questions/29020081/Multiple-Voice-Vlans-on-Cisco-switch.html
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How does DHCP work with VoIP phones?  Do they phones need to get restarted every time the DHCP lease expires?  Or it works no different than a regular computer client?
For the first time phones need to be booted to obtain IP addresses.  Also when DHCP or DNS server changes.  What about a regular DHCP refresh at the end of the lease duration?
We use Polycom ip phones over poe.
Please advice.
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Avaya IP Office 500 switch/hub going into a patch panel. - All seems OK

Running Avaya  IP Office 8.1 Manager on the PC. Our phone system is no longer greeting callers. It just rings and rings and rings.
I'm able to ping the gateway and static ip. Everything was working up until a few days ago apparently.

I tried rebooting the Avaya and also the IP Office 500 Switch.

Any ideas or direction is greatly appreciated.

TY All...
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I want to make a Call Manager unavailable for CIPC registration. What services would I need
to shut down so that any CIPC that attempts to connect goes to the secondary in its
configured list of CUCM servers?
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I've been tasked with the possibility of upgrading our offices phone system.
Our current phone system runs off CAT3. Newer systems CAT5 or better. Our network is pretty old. I don't have CAT5 everywhere there is a phone and not even everywhere there's a computer. Those users end up on wifi. Anyway, from what I gather we can use a passthrough on the phones to connect the phone then the computer back to the network. We've got about 55 users and about the same amount of phones in our main office. Our plan was to also roll out VPN phones to multiple locations as we have jobsite trailers all over several states.

Is it advisable to use the passthrough on these phones? Or is having a dedicated line going to be significantly better? We also have a few places in the building where I've had to add a few small switched and I'm concerned we'll get some call degradation because of this.

Like I said, our network is a bit older and a lot of it has been band-aid fixes by adding some wifi AP's and daisy-chaining a few small switches in some areas for additional personnel over the years.

Personally I'd love to have our building rewired but I'm not totally sure it's necessary, however I don't want to cause more problems or increase load on a network that's barely adequate as it is.

If you need more info from my I'd happily provide it. I'd like some advice from anyone else who's done similar upgrades or implemented phone systems on current networks.


Thanks
0
Hello,

Could someone help resolve issue where our SIP dial peer is ignored/mismatched and explain how to fix it without detriment to the existing call routing???
 
THE SETUP

We have an existing UK office collab setup which consists of:
 
CUCM (9.1) <-> ISR2921 (15.2(4)M4) <-> PSTN (ISDN30)
 
Currently all calls are routed from the CUCM to H323 gateway then out to PSTN.
We are in the process of testing SIP as a possible way forward (to replace PRI connection at some point).
 
I have setup everything in terms of SIP there is to be configured (as far as I can tell).
I have also configured route pattern on Call Manager to test calls to my mobile.
 
Calls come from CUCM to h323 gateway, get matched by multiple dial peers but in the end are send out the old PSTN connection. I am attaching debug dialpeer result for my test call -> test_call_debug.txt for anyone interested enough to take a look.
 
Here is my voip/sip config:

voice service voip
 ip address trusted list
  ipv4 172.0.0.0
  ipv4 46.165.252.40
  ipv4 5.150.254.205
  ipv4 72.251.241.166
  ipv4 83.222.249.39
  ipv4 54.172.60.1
  ipv4 54.172.60.0
  ipv4 54.172.60.2
  ipv4 54.172.60.3
  ipv4 35.156.191.128
  ipv4 35.156.191.129
  ipv4 35.156.191.130
  ipv4 35.156.191.131
 address-hiding
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 

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Does skype limit callers to 50 calls per day

https://voicent.com/kb/index.php/general/370/skype-limits-outbound-calls-to-50-a-day

This page is old 2009; maybe skype allows more now

I dont know how to test because I call less than 10 per day
0
I've got a NETGEAR  GS724T V4 Managed switch with a VLAN set up using the Auto VoIP setting for 6 ports to serve VoIP phones at a client's location. The other ports are set to default. The IP for the switch is 192.168.1.200.  The switch is on a 192.168.1.1 network.


1) Do I need to set up another subnet for this VLAN (192.168.2.1), or will these ports just be isolated from the traffic on the other ports of this switch.
2) If I do need to set another subnet, how do I set either static IPs or DHCP for these 6 ports on the subnet?

Thanks lots for your insight.
0
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microsoft edge browser windows 10

I highlight a phone number (sequential digits) and there WERE choices for open with.

I chose skype

and want to change my choice

but my choice options are now gone
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Does Cisco CUBE pair need to have IP Addresses in 2 different subnets from MPLS CE Routers? We have a pair of CEs connecting to the Cisco CUBE for the voice. Please clarify.

Thanks;
0
What is the Cisco term of the DX80 telepresence bridge point? Cisco Spark connect point?
0
I work for a company in which the general manager has VOIP phone, Windows computer and iPhone devices.   The general manager has a list of contacts that he would like to have automatically synchronize to all of these devices.  Where would be the best place to edit this contact list and get it to synchronize to all of these devices automatically?  If there is a better process to do this I would appreciate input.  Thanks.
0
i'm trying to add voice control to ubuntu 17.10.  i saw a video where the command "v-c" was added to ubuntu and somehow you could write code with it.  it says to write v-c<<help (i think) .  i tried to get help and my computer sayts the command does't exist
0
We’re using a SonicWall TZ-215 firewall.  Our LAN X0 port is setup as 192.168.0.1—254, with 192.168.0.2 setup as the gateway, 255.255.255.0 as the subnet.  

We’re getting VoIP phones, and the vendor is setting them up with static addresses between 192.168.1.100—200, subnet 255.255.255.0.  How do I configure the SonicWall to get the phones to access the 192.168.0.2 gateway?  (X3—X6 interfaces are unused on the SonicWall, if needed for the solution)
0
When I use The Microsoft Connectivity Analyzer for Skype for business connectivity I get an error when it is attempting to obtain the SSL certificate from remote server. please see attached.
C--Users-ogurekm-Documents-SSLSkypeE.pdf
0
We currently have two physical incoming telco lines - one VOIP and one currently ISDN.  The VOIP line is currently being used for all data traffic.  The latter is now being "force" converted to VOIP by the telco provider, and I was advised we would need a second router for this telco line.

My question is, whether it is possible to double the bandwidth by running data traffic over both lines (no specific requirement as to what traffic runs over which telco line) in parallel?  Most likely there would be problems with colliding  IP addresses - default gateway and any other IP addresses utilized by the routers themselves.  I imagine there are routers capable of addressing two incoming telco lines, but how expensive would one be im comparison to two separate commonly available routers?

Any suggestions would be appreciated.  The other posts on this topic all address a different scenario.
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A client has a VOIP phone system, and a Comcast gateway. On the gateway, one cannot turn off  SIP/ALG, which makes the VOIP  calls inconsistent.  They have an unmanaged switch now that cannot be programmed to remove the SIP/ALG disturbance.  What I need to a 24 port gigabit switch that can be programmed to remove this SIP/ALG  setting and produce VLANS to segment the phone and other traffic.

I'm wondering budget-wise whether the Netgear JGS524E would allow me to turn off SIP/ALG as well as create the necessary VLANS.  If not, can you recommend one?
0
I am trying to do an assessment for skype for business online using MS tool NetworkAssessmentTool.exe
and then ResultsAnalyzer.exe. But when I try to run the results tool I get an error about the delimiter.
See below. Is there a switch that needs to be run when executing the NetworkAssessmentTool or
the ResultsAnalyzer tool? I looks like the results file that that was generated is largely tab delimited.

C:\MSSKYPE_Assess>ResultsAnalyzer.exe \Users\LeeRoyJenkins\AppData\Local\Microsoft Skype for Business Network Assessment Tool\performance_results.tsv
Delimiter must be a single character.

I tried changing the delimiter from <tab> to single quotes. But I get the same error. Any thoughts?

https://blogs.technet.microsoft.com/skypehybridguy/2017/08/11/assess-your-networks-readiness-for-skype-for-business-online/
0
skype classic latest edition for windows 10
I dont want to install skype on windows store as an app

For years on windows I have been trying skype click to call
On some computers it works for a small amount of time

Now I want to give up and stop trying. Takes too much time. I have been trying for years. I dont want to uninstall browsers/skype and restart my computer for hours just to make 5 phone calls

click to call skype on firefox does not work

but now I am stuck with something that is worse

I can click on phone number to activate skype but skype does not make phone call. Skype opens and does not do anything.


+1 phone number that I can not highlight is worse than what I started with.
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CONTEXT:  We have all layer-3 switches.  At present, we are only concerned about VoIP and VIDEO QoS.  We believe configuration simplicity outweighs complexity if complexity is not needed to meet our needs and wants

QUESTION:  What is better QoS: DSCP or IPP ?  Why?
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We are looking for soft phone system for a small contact center.
and prefer to use Twilio as phone carrier. any suggestion which company or software we can use?
0
I have a Cisco UCM voip system that I am admittedly a rookie at.  We recently went through a reorganization of equipment and six of our fax machines were moved to different locations within the building.  We were trying to get off POTS lines anyway so we decided to purchase Cisco ATA 190 devices so that our Ricoh MFP devices could participate on the digital phone system.  These installed just fine and have worked fine, or so we thought.  After being on them for a month or two I am getting a lot more help desk calls stating that incoming faxes are taking forever or failing altogether.  For example, I have a site that faxes here quite often.  When they fax it will often times take 20 minutes to receive a fax from them successfully when the fax machine here isn't busy doing anything else and the other side is not busy either.  In another instance, I had a different facility report that they could not fax to us.  I happened to have the speaker turned on in that Ricoh MFP and I could hear it pick up and try to negotiate and then it would drop the call.  The process would repeat dozens of times and then a fax would finally come through.

I am curious if there are any settings in the ATA 190 that I need to tweak or if there is something that needs to be tweaked in Cisco UCM for the device?  Nothing in the interface on the ATA 190 sticks out at me and I don't even see a setting to lower the baud rate.  I did set up the ATA to send debug messages to a syslog server so maybe that …
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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).