Diagnosing Audio Problems from the Yealink IP Phone
Poor voice quality, when clients calls in we can hear them clearly but they can not hear us. The same happens when we are dialing out, we can hear them them but they can not hear us.
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Various potential network problems might be causing the problem. Access the Network Audio Quality screen. In the event you experience poor audio quality during a call, the problem is likely with your
network. This screen provides information that can help identify causes of poor audio quality.
Note:
Note: The Network Audio Quality button displays only when you are on a call
(off-hook).
1. Select Network Audio Quality from the Options Main screen (accessed by pressing the
phone's Option button).
If administered, the Audio Status screen displays.
2. To return to the first Options Main screen, press the Return softkey.
You should remember, this is a fee service, so the quality is going to eb a little poor. Some of the things to increase, would be to eliminate all other softwares from running that would use the wifi or network connection. And have the person on the other end do the same. Of all the IM pc to pc chat/talk programs, Yahoo, seems to have the worst quality. if it is for voice only, I might suggest Gtalk, it seems to have the quality consistantly.
I have had the same problem.... When the video was played after recording no or very bad quality sound were herd.... The solution is to do all of the following 1)Changing the cover 2)Changing the fones body 3)Changing both the Speaker and the microfone....
The handsets will eventually go bad. If it is every single handset that you have, the problem is with the system. But if the problem is only with a few of your phones, you might consider replacing the handsets. We at IdacomUSA.com have replacement Cisco handsets for sale at only $25. Hope this helps you out :)
You may be behind a firewall or NAT. You can set an option that you are; if that still does not work, you may need to open some ports on the firewall.
Yahoo! Messenger uses a variety of ports. ServicePorts
Chat & Messenger
TCP Port 5050: Client Access only
Insider/Room Lists
TCP Port 80: Client Access only
File Transfer
TCP Port 80: Server Access. Your ISP may block this port, as its used for web hosting. You can change port in Messenger, Preferences, File Transfer.
Voice Chat
UDP 5000-5010 TCP 5000-5001: Client Access If UDP Fails, TCP will be used instead, see below.
WebCam
TCP Port 5100: Client Access
Super Webcam
TCP Port 5100: Server Access
P2P Instant Messages
TCP Port 5101: Server Access PMs between Buddys may not use the Yahoo! Server, but this is not a requirement.
1st check if your fone is ringing, coz if not u cant hear it , and it will continue to ring and after that it will goes to your voice mail if unanswered
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.
Set up the callers in your contacts. You must have the area code for the name to show up. You must dial the area code for the name to show up when dialing out.
The hearing problem could be the person you are talking to, if your volume is all the way up. If you are sure that it is you I would talk to a motorola technician. Let me know how it goes.
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