This section addresses common problems that you may experience with Cisco Unified IP Phone s, gateways, and related devices. You may experience voice-quality issues including lost or distorted audio signal during phone calls. This section covers some common voice-quality problems. Common problems include audio breaks like broken words or the presence of odd noises and audio distortion, such as echo, and watery or robotic voice quality.
One-way audio, that is, a conversation between two people where only one person can hear anything, does not actually represent a voice-quality issue, but this section covers this issue.
One of the most common problems that you may encounter involves broken audio signal often described as garbled speech or lost syllables within a word or sentence. Packet loss means that audio packets do not arrive at their destination because they were dropped or arrived too late to be useful. Jitter describes the variation in the arrival times of packets. Notice that this is not the time that it takes for a packet to get from point A to point B but is simply the variation in packet arrival times.
Many sources of variable delay exist in a network. You can control some of these but not others. You cannot entirely eliminate variable delay in a packetized voice network. Digital Signal Processors DSP on phones and other voice-capable devices by design buffer some of the audio in anticipation of variable delay. This dejittering occurs only when the audio packet reaches its destination and is ready to be put into a conventional audio stream.
Because the jitter buffer is adaptive, if a burst of packets is received, the Cisco Unified IP Phone model can play them out in an attempt to control the jitter.Jabber Voice Mail Demo
The network administrator needs to minimize the variation between packet arrival times by applying quality-of-service QoS and other measures in advance especially if calls cross a WAN. Some video endpoints may not support G. Use another codec, such as G. Remember that delay by itself will not cause clipping, only variable delay.
Notice in the following table, which represents a perfect trace, the arrival times between the audio packets which will have an RTP header will be 20 ms. In a poor quality call such as a call with a lot of jitterthe arrival times would vary greatly. The following table illustrates a perfect trace.
Placing the packet analyzer into various points in the network will help narrow the number of places from which the delay is coming. If no analyzer is available, you will need to use other methods. Examine interface statistics of each device in the path of the audio. Audio problems occur while a call is in progress. Devices, where a higher speed interface feeds into a lower speed interface, provide the most common sources for delay and packet loss.
If the poor audio quality occurs only when communicating to the remote site, the most likely causes of the problem include. On the LAN, the most common problems represent physical-level errors such as CRC errors that faulty cables, interfaces, or by incorrectly configured devices such as a port speed or duplex mismatch cause.The document defines a vocabulary that can be used to discuss symptoms of voice quality problems.
Sound files are included to aid in the process of identification of the symptom. Also included where possible are one or more common causes not necessarily the only ones for the symptom that is defined. The sound files and names of symptoms used in this document are based on common language used in Cisco Technical Support service requests, on the Technical Support website, and other sources.
This document is intended to be a living resource in that the symptoms listed are expected to be revised as new problems arise and additional recordings become available. This is the suggested high level procedure to troubleshoot voice quality problems, in conjunction with this document:.
Check the sound files in this document for a symptom that matches or resembles the one that is experienced. You might wish to provide your users with a link to this document if you have not personally heard the symptom. Access the Cisco Support Community in order to research the problem or ask questions. If no resolution is gained by use of the Cisco Support Community, make use of the symptoms vocabulary defined in this document in order to raise a Technical Support service request.
The Technical Support engineer might ask you to make use of a Cisco utility that allows you to capture the Real Time Protocol RTP stream of the problem and convert it to a. If you agree, an appropriate portion of the wav file can be used in this document and referenced from the TAC CC so that others can share the benefit of your experiences. These definitions were developed and applied in order to categorize the voice quality problem symptoms:. This is typically any noise on the line or in a voicemail message in addition to the voice signal.
Noise typically leaves the conversation intelligible but still far from excellent. Static, hum, crosstalk, and intermittent popping tones are examples where the calling and called parties can understand each other, but with some effort. Some noises are so severe that the voice becomes unintelligible. One such example, among the samples provided in this document, is a motor sound.
It can be heard at either end of the call, in varying degrees and with many combinations of delay and loss within the echoed signal. On some occasions, the voice becomes unintelligible. Note : The categorization of the symptoms is to a large degree dependent on the severity of the symptom, perceptual factors, and cultural factors. Therefore, the placement and grouping of symptoms within categories is in many cases arguable.
In addition, there can be situations where the categories overlap. For example, static on the line can cause some form of voice distortion. This is a best attempt to give some structure to these terms and define the vocabulary.
In this section, you can listen to sound recordings of the symptoms defined, along with control samples that allow you to hear the same recording without the accompanying symptom. A snippet sample of the symptom is included in order to allow for quicker download times and easier browsing. The full recording provides a longer sample so that the symptom can be properly heard.
The symptom recordings are kept as MP3 files and can be played by any sound player that supports the MP3 file format. Also, included where possible, are one or more common causes not necessarily the only ones for the symptom that is defined. Note : Remember to keep your initial volume settings low. Increase volume as needed once you are comfortable with the volume levels of the recordings.
If you have technical difficulties when you listen to or download these recordings, see the Common Problems Hearing Sound Files section of this document.One of the most important steps in troubleshoot voice quality related issues is isolate them, either to a particular phone, set of phones, switch, gateway, etc. Once the issue has been identified via users that report issues, Call Detail Records CDRs or some other means, it is important to gather data to help isolate it.
Step 1. Step 2. Take note of the physical location ex. Site A, Floor 2user name of the user's the phonedirectory numbers DNsphone model exphone firmware ex.
Create a spreadsheet with this information sorted by physical location. Step 3. The audio for external and internal calls typically takes different paths. Verify that the phone with the issue runs the same firmware as other known phones that work fine, if the firmware is different a firmware upgrade can resolve the issue.
If the audio issues are still present, there is likely an issue with the physical phone. Create a simple topology that shows the path that the RTP packets takes. Once you have the topology written out, the first step is to take packet captures on one side of the topology and work your way to the other end of the topology.
Use Wireshark to decode the RTP stream and play back the audio. If there is no issue with the audio the users voice is clearyou can eliminate the phone, cabling from the phone to the switch, and the phone equipment handset, headset, speakerphone as the source of the poor quality. At this point move to Step b if there is no issue with the audio.
Recognizing and Categorizing Symptoms of Voice Quality Problems
If there is no issue with the audio move to Step ccontinue to gather packet captures along the RTP path. If there is no issue with the audio move to Step d to gather more packet captures. The next packet capture must be taken from the GW. If there is an audio issue with audio quality at egress, you have isolated the issue to the GW. Collecting a packet capture from a Cisco IP Phone.
How to troubleshoot voice quality issues in a UCM environment bad sound, no audio. Skip to content Skip to footer. Available Languages. Download Options. Updated: May 25, Contents Introduction. Prerequisites Requirements Cisco recommends that you have knowledge of these topics: Cisco Unified Communication Manager.
Voice over IP VOIP Components Used The information in this document is not based on any specific software or hardware versions: Background Information One of the most important steps in troubleshoot voice quality related issues is isolate them, either to a particular phone, set of phones, switch, gateway, etc. Where to Start? Questions to ask in all Scenarios These questions helps to narrow down the voice path of the effected calls.
Does the issue occur on only external calls, only internal calls, or both? Additional Resources 1. Collecting a packet capture from a Cisco IP Phone 2.
How to troubleshoot voice quality issues in a UCM environment bad sound, no audio 4. Contributed by Cisco Engineers Jason Wiatr. Was this Document Helpful? Yes No Feedback. Related Cisco Community Discussions.I am having an audio related issue with Jabber. While I work I usually listen to music over headphones. I have Jabber set to use an audio notification when I get messages. When receiving a new Jabber message while using headphones, for some reason Jabber defaults to using the built in laptop speakers and NOT my headphones.
To make things worse, while the jabber tone is playing it also redirects all other streaming audio such as my music to the internal speakers so for seconds everyone around me can hear the jabber tone and my music. After the jabber tone is done the streaming audio goes back to my headphones. I have noticed that when this happens briefly I will see two audio devices listed when I click my volume control icon.
It seems as if Jabber is defaulting to the built in speakers audio device despite the headphones being connected as the primary device.
I am using Windows 7 and the latest audio drivers. The headphones are 3. To get around this on my previous laptop I had to disable all Jabber sounds. It seems I will have to do the same with this until the bug is fixed. I also wanted to note that I never had this issue when using WebEx Connect.
It started once we switched to Jabber at my office. And try to adjust the Audio Device in Jabber. I see your suggestion and while this will do as a work-around until a real fix is implemented this creates another problem. On days I am in the office I want it to default to headphones. On days I work from home I want it to default to the laptop's speakers. I don't want to have to change this option multiple times each week.
Is there no way for Jabber to just use the default in-use device like any other application with audio can do? My question: can I select to different devices for audio out, so that the user can hear the softphone ringing the warning sounds via the pc-speaker and then, after he has picked up the phone the sound comes only from the headset? Late post on this but you can set the defaults in Jabber by going to options - audio - advanced option and then set your preference order.
Buy or Renew. Find A Community. We're here for you! Turn on suggestions. Auto-suggest helps you quickly narrow down your search results by suggesting possible matches as you type.For Packet Voice to be a realistic replacement for standard public switched telephone network PSTN Telephony services, the received quality of Packet Voice must be comparable to that of basic telephone services.
This means consistently high-quality voice transmissions.
Like other real-time applications, Packet Voice has a wide bandwidth and is delay sensitive. For voice transmissions to be intelligible not choppy to the receiver, voice packets cannot be dropped, excessively delayed, or suffer varying delay otherwise known as jitter. This document describes various Quality of Service QoS considerations that help troubleshoot choppy voice issues. The main reasons for choppy voice problems are lost and delayed voice packets.
Basic understanding of voice prioritization, fragmentation, different codecs and their bandwidth requirements. The information in this document applies to all Cisco voice gateways software and hardware versions. The information presented in this document was created from devices in a specific lab environment. All of the devices used in this document started with a cleared default configuration. If you are working in a live network, ensure that you understand the potential impact of any command before using it.
For more information on document conventions, refer to the Cisco Technical Tips Conventions. Choppy voice quality is caused by voice packets being either variably delayed or lost in the network. When a voice packet is delayed in reaching its destination, the destination gateway has a loss of real-time information. In this event, the destination gateway must predict what the content of the missed packet can possibly be. The prediction leads to the received voice not having the same characteristics as the transmitted voice.
This leads to a received voice that sounds robotic. If a voice packet is delayed beyond the prediction capability of a receiving gateway, the gateway leaves the real-time gap empty. With nothing to fill up that gap at the receiving end, part of the transmitted speech is lost. This results in choppy voice. Many of the choppy voice issues are resolved by making sure that the voice packets are not very delayed and more than that, not variably delayed.
Sometimes, voice activity detection VAD adds front-end clipping to a voice conversation. This is another cause of choppy or clipped voice. The various sections in this document show how to minimize the instance of choppy voice. Most of these measures require assuring the introduction of minimum jitter in your voice network.
Before you consider applying any measures for minimizing jitter, provision sufficient network bandwidth to support real-time voice traffic. For example, an 80 kbps G.
The bandwidth requirements vary based on the codec used for compression. Different codecs have different payloads and header requirements. Usage of VAD also affects the bandwidth requirement. For example, the bandwidth required for a voice call using the G. This yields If you cannot hear the other participant smake sure your loudspeakers or headphones are connected.
Then check all volume controls:. If this does not solve the problem, it may be that audio is not being sent from the other end. Ask the other participant s to perform the microphone check described below. If the other call participant s cannot hear you, make sure your microphone is properly connected and not muted.
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Troubleshooting QoS Choppy Voice Issues
Jabber Video: Troubleshoot audio issues. Labels: WebEx Telepresence. You cannot hear others If you cannot hear the other participant smake sure your loudspeakers or headphones are connected. Then check all volume controls: Your headphones or loudspeakers may have their own volume buttons or switches.
Cisco Jabber Video has its own volume setting on the [x-ref] Using the pop-up toolbar [p. Others cannot hear you If the other call participant s cannot hear you, make sure your microphone is properly connected and not muted.
Low or distorted sound If call participants are experiencing distorted sound, very low sound, or echos: Check whether any of you have enabled microphone boost, echo cancellation, gain control, noise reduction, digital effects, or similar features for your audio devices.
Troubleshoot Voice Quality Issues
Turn all such audio device features off for Jabber Video to work optimally. Latest Contents. My client just informed me that he has RV site to site setup assistance. Created by Dave on AM. As a result of the pandemic, I've been thrown into setting up a site-to-site VPN.Introduction Really understanding the problem Collecting additional details Narrowing down to a specific situation Troubleshooting Tools.
In this guide I will try to describe a structure to use when troubleshooting voice issues like choppy or clipping sound, hissing noise, dead air or silence, one way audio, and others, rather than provide merely technical recommendations that may fix those.
The goal here is to add another tool when you have a TAC case open so to provide us with more information, or even before opening the case to attempt a quick narrow down of the problem, and maybe a fast solution as well. The first thing we need to do, is to take the problem description to a concrete, short statement. Just stating that "the sound is bad", or that "there are voice quality issues" won't help you or your TAC engineer conclude a thing. Those statements however, reflect what end users may experience, and it is ok for them to express that way, the person working on the problem is the one in charge to gather a clear understanding of the situation.
Before even moving to the part where we start collecting details, lets ask the following questions:. Once we answer those, we will know what we are after later when checking configurations and logs. We will end up with a statement such as:.
Now is time to fully understand the topology in which the problem is happening. Audio streams are just like any other data streams in the sense that they flow through a cable or through wireless networks from one device to another. Having said that, the first thing to do after the problem statement is ready, is to narrow down probabilities to as few components as possible.
Such task can only be done when there is a clear topology outline. Answer the following sample questions:. Does it happen with calls from one IP Phone to an outside number only? Does it happen when connecting to a internal voice service like a contact center or voicemail system? At this point I would like to clarify myself when saying that voice or video streams " flow through a cable or through wireless networks from one device to another ".
There are 2 perspectives to take into account. Voice packets flow between phones, switches, routers, transcoders and other devices as bits, so every router, cable, switch and telephony device is suspect of dropping packets or causing delays; that would the transport perspective. From the signaling perspective, you must remember to always understand which devices are also voice devices, for example transcoders, Media Termination Points, Trusted Relay Points, RSVP agents, audio proxys and so forth.
Why do I make this differenciation? Easy, take a look at the following diagram:. Following the blue dotted lines you see how the RTP with the real audio packets flow from one device to the other.
It helps us determine which layer 1 through layer 3 paths are taken to deliver the information, and would help us focus on which devices to troubleshoot, however, lets imagine the topology from above is not telling us the entire story. Our topology now transforms into this:.