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VOIP works

At its best basal level, VOIP artlessly break bottomward calls and encodes them into packets that can again be transmitted over the internet. When they get to the advised recipient, they are again reassembled into sounds for the listener.

The affection of the complete will depend on the codec's acclimated to encode the sounds and factors such as the affection of the connection.IP phones can now be begin in abounding forms.

There are ATA phones that assignment through a affiliation amid the net and absolute blast jacks, those that affix through any Wi-Fi or Ethernet access and softphones that assignment throughcomputer application installed on a computer.

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fax machine does not receive FAX


The problem is that the codecs used by VoIP IADs are designed to compress voice, not the analog signals sent and received by modems.
In a VoIP Internet world, voice is first converted into packets and then they are sent over the connections that make up our our vast Internet. They may take slightly different times to arrive at their destination. In doing so some packets may be discarded, but the end result is that the receiving VoIP device has enough packets to make a clear and understandable conversation.
We suggest these settings on a fax machine for faxing over VoIP; slowing the transmission rate down and allowing the machine to continue receiving the transmission even though a few bits of data were lost, then faxing over VoIP can become more consistent. Our suggestions in many cases can resolve issues that prevent faxing over a VoIP connection, but not in all cases. If after trying and making all the VoIP fax changes we suggest you still cannot fax over your connection then try a Internet Fax service.

Aug 10, 2011 | HP Office Equipment & Supplies

1 Answer

My problem is that I have just purchased Vonage as a VOIP phone line. I have a HP LaserJet 3055 printer/fax and I am trying to send faxes but it does not work. I was instructed to lower the baud rate, however I have no idea how to do this. when i go to fax setup->all faxes>V.34-> it can only go on and off. I need to know how to change this setting asap


Configuring the V.34 setting to Disable (Enable is the default setting) to lower the speed.


Voice over IP (VoIP) services provide normal telephone service, including long distance service through a broadband Internet connection. These services use packets to break up the voice signal on a telephone line and transmit it digitally to the receiver, where the packets are reassembled. The VoIP services are often not compatible with fax machines.

If you continue to have problems faxing, contact your VoIP provider. Because the installation process varies, the VoIP service provider will have to assist in installing the all-in-one fax component.

Although a fax might work on a VoIP network, it can fail when the following events occur:

• Internet traffic becomes heavy and packets are lost.

• Latency (the time it takes for a packet to travel from its point of origin to its point of destination) becomes excessive.

Feb 15, 2011 | HP LaserJet 3055 All-In-One Printer

4 Answers

how would fax run on VOIP? what requirment it needs?


typically faxing doesn't work on VOIP. i tried many times and sometimes it would go through although it said it didn't. other times it didn't work at all. VOIP has a limited bandwidth and the fax needs to send more info over the line than VOIP makes room for.

to have the best luck, stop all downloads and other network traffic while attempting to fax, so that there are no other programs or devices using the internet. the fax will have its best chance of working that way, although it probably still won't

here are some very good tips as well:
http://chris.pirillo.com/how-to-fax-over-voip-on-the-internet/

Feb 18, 2009 | Lexmark X4270 All-In-One InkJet Printer

1 Answer

Linksys adapter


Factors that Affect Voice Quality
  • Audio Compression Algorithm
    Speech signals are sampled, quantized and compressed before they are packeted and transmitted to the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per second with 12-16 bits per sample. The compression algorithm plays a large role in determining the Voice Quality of the reconstructed speech signal at the other end. The SPA supports the most popular audio compression algorithms for IP Telephony: G.711 a-law and µ-law, G.726, G.729a and G.723.1. The encoder and decoder pair in a compression algorithm is known as a codec. The compression ratio of a codec is expressed in terms of the bit rate of the compressed speech. The lower the bit rate, the smaller the bandwidth required to transmit the audio packets. Voice Quality is usually lower with lower bit rate. However, Voice Quality is usually higher as the complexity of the codec gets higher at the same bit rate.
  • Silence Suppression
    The SPA applies silence suppression so that silence packets are not sent to the other end in order to conserve more transmission bandwidth. Instead, a noise level measurement can be sent periodically during silence suppressed intervals so that the other end can generate artificial comfort noise that mimics the noise at the other end using a CNG or comfort noise generator.
  • Packet Loss
    Audio packets are transported by UDP which does not guarantee the delivery of the packets. Packets may be lost or contain errors which can lead to audio sample drop-outs and distortions and lowers the perceived Voice Quality. The SPA applies an error concealment algorithm to alleviate the effect of packet loss.
  • Network Jitter
    The IP network can induce varying delay of the received packets. The RTP receiver in the SPA keeps a reserve of samples in order to absorb the Network Jitter, instead of playing out all the samples as soon as they arrive. This reserve is known as a Jitter Buffer. The bigger the Jitter Buffer, the more jitter it can absorb and the bigger the delay it can introduce. Therefore the jitter buffer size should be kept to a relatively small size whenever possible. If jitter buffer size is too small, then many late packets may be considered as lost and thus lowers the Voice Quality. The SPA can dynamically adjust the size of the jitter buffer according to the network conditions that exist during a call.
  • Echo
    Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near-end echo. The SPA has a near end echo canceller with at least 8 ms tail length to compensate for impedance match. The SPA also implements an echo suppressor with comfort noise generator (CNG) so that any residual echo will not be noticeable.
  • Hardware Noise
    Certain levels of noise can be coupled into the conversational audio signals due to the hardware design. The source can be ambient noise or 60Hz noise from the power adaptor. The SPA hardware design minimizes noise coupling.
  • End-to-End Delay
    End-to-end delay does not affect Voice Quality directly but is an important factor in determining whether subscribers can interact normally in a conversation taking place over an IP network. Reasonable delay figure should be about 50-100ms. End-to-end delay larger than 300ms is unacceptable to most callers. The SPA supports end-to-end delays well within acceptable thresholds.

Dec 01, 2007 | Linksys VOIP VONAGE PHONE ADPTR 2-PORT...

1 Answer

Communication problems over IP phone


I have found many possible causes. Firmware Levels Outdated firmware in routers, VOIP phones, Firewalls, etc. can cause one-way audio. Ensure you have the latest updates for all the devices in the call path. Configuration Particularly if NAT is involved in the call path, configuration of the various devices may be a problem. Check to see if all devices are configured appropiately for your envioronment. Has anything changed, carriers harware, quos settings on router ? packet loss to these locations should be tested @ time it happens most, don't use it as gospel retur, ttl and re-xmits' do same.. Bob Finding the Cause The basic troubleshooting technique is to use a tool like Ethereal to capture SIP and RTP packets at each point in the call path where packets could be lost. Interperting the resulting captured packets requires some familarity with how networking and VOIP work. For example if the call path is: VOIP phone/device ---- firewall ---- sip proxy ---- firewall ---- asterisk Troubleshooting Steps Start capturing at point Make a VOIP call that will have one-way audio Analyze capture If problem found, fix and retest Otherwise move capture point to the next point (a, b, c, d, etc) and start over If the problem is intermittent, then a long term simultanous capture at multiple points can be used to attempt to capture a comple call with the problem. Most capture tools will let you capture only traffic from selected devices, so the volume of captured information can be kept to a reasonable size. If a back-to-back SIP user agent (for example a Session Border Controller ) is part of of the call path, then it may be necessary to capture all VOIP traffic at some points to ensure catching the wanted call since the IP addresses can change when traversing this device. Resources How To Debug and Troubleshoot VOIP

Nov 14, 2006 | Nortel I2002 IP Phone w/Text Keycaps, No...

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