Voice Quality
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.
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Gateways
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Phones
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Networks
Lost or Distorted Audio
Symptom
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). Two common causes for this exist: packet loss and/or jitter. 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. In the ideal situation, all Voice over IP (VoIP) packets would arrive exactly at a rate of 1 every 20 microseconds (ms). 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.
Possible Cause
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.
The Cisco Unified IP Phone model 7960 can buffer as much as 1 second of voice samples. Because the jitter buffer is adaptive, if a burst of packets is received, the Cisco Unified IP Phone model 7960 can play them out in an attempt to control the jitter. 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.728, and using G.728 may result in noise. Use another codec, such as G.729.
Recommended Action
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When you are faced with a lost or distorted audio problem, first try to isolate the path of the audio. Try to identify each network device (switches and routers) in the path of the call audio stream. Keep in mind that the audio may be between two phones, or between a phone and a gateway, or it could have multiple legs (from a phone to a transcoding device and from there to another phone). Try to identify whether the problem occurs only between two sites, only through a certain gateway, on a certain subnet, and so on. This will help narrow the number of devices that you need to look at more carefully.
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Next, disable silence suppression (also known as Voice Activation Detection or VAD). This mechanism does save bandwidth by not transmitting any audio when silence occurs, but may cause noticeable or unacceptable clipping at the beginning of words.
Disable the service in Unified Communications Manager Administration and choose
. From there, choose the server and the Cisco CallManager service. -
Set SilenceSuppression to False to disable for all devices in a Cisco Communications Manager cluster; alternatively, you can set SilenceSuppressionForGateways to False. When in doubt, turn both off by choosing the value False for each.
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Using a network analyzer, if a network analyzer is available, check whether a monitored call between two phones has 50 packets per second (or 1 packet every 20 ms) when silence suppression is disabled. With proper filtering, you can identify whether an excessive number of packets are lost or delayed.
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 jitter), the arrival times would vary greatly.
The following table illustrates a perfect trace.
Packet Number |
Time - absolute (sec) |
Time - delta (ms) |
---|---|---|
1 |
0 |
|
2 |
0.02 |
20 |
3 |
0.04 |
20 |
4 |
0.06 |
20 |
5 |
0.08 |
20 |
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.
Diagnostic Call Detail Records (CDR) specifies another tool for tracking calls with poor voice quality. Refer to the Call Reporting and Billing Administration Guide for Cisco Unified Communications Manager for more information about CDRs.
Correcting Audio Problems From the Cisco Unified IP Phone
Symptom
Audio problems occur while a call is in progress.
Possible Cause
Devices, where a higher speed interface feeds into a lower speed interface, provide the most common sources for delay and packet loss. For example, a router may have a 100-Megabyte (MB) fast Ethernet interface that is connected to the LAN and a slow frame-relay interface that is connected to the WAN. If the poor audio quality occurs only when communicating to the remote site, the most likely causes of the problem include
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The router was not properly configured to give voice traffic priority over data traffic.
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Too many active calls exist for the WAN to support (that is, no call admission control restricts the number of calls that can be placed).
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Physical port errors occur.
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Congestion in the WAN itself occurs.
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. Make sure that the traffic is not crossing any shared-media device, such as a hub.
Recommended Action
The Cisco Unified IP Phone model 7960 provides another tool for diagnosing possible audio problems.
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On an active call, you can press the i or? button twice rapidly and the phone will display an information screen that contains packet that receive and transmit statistics, as well as average and maximum jitter counters.
Note
On this window, jitter represents the average of the last five packets that arrived; the maximum jitter designates the maximum for the average jitter.
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Situations could also occur where the traffic is taking a slower path through the network than expected. If QoS is configured correctly, the possibility exists that no call admission control exists. Depending on your topology, you can accomplish this through the use of Locations in Cisco Unified Communications Manager Administration configuration or by using a Cisco IOS router as a gatekeeper. In any case, you should always know the maximum calls that are supported across your WAN.
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Crackling represents another poor-quality symptom, which a defective power supply or some kind of strong electrical interference close to the phone sometimes causes. Try swapping the power supply and moving the phone.
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Verify gateway and phone loads. at www.cisco.com for the latest software loads, new patches, or release notes that relate to the problem.
After you apply the appropriate fix, verify the sound quality by performing the following procedure:
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Test by disabling silence suppression; then, place calls between the two sites. Do not place the calls on hold or on mute because this will stop packets from being transmitted.
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With the maximum number of calls across the WAN, the calls should all have acceptable quality.
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Test to make sure that a fast busy is returned when you try to make one more call.
Echo
Symptom
Echo occurs when the speech energy that is being generated and transmitted down the primary signal path gets coupled into the receive path from the far end. The speaker then receives his or her own voice, delayed by the total echo path delay time.
Voice can reflect back. This can happen but goes unnoticed in a traditional voice network because the delay occurs so lowly. To the user, it sounds more like a side-tone than an echo. In a VoIP network, it will always be noticeable because packetization and compression contribute to the delay.
Possible Cause
Remember that the cause of the echo always lies with analog components and wiring. For instance, IP packets cannot simply turn around and go back to the source at a lower audio level or on digital T1/E1 circuits. The only exception may occur if one party is using a speakerphone that has the volume set too high or other situations where an audio loop is created.
Recommended Action
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Make sure that the problem phones do not use the speakerphone and that they have the headset volume set to reasonable levels (start with 50 percent of the maximum audio level). Most of the time, the problems occur when you attach to the PSTN by way of a digital or analog gateway.
Testing the Gateway
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Determine which gateway is being used. If a digital gateway is in use, you may be able to add additional padding in the transmit direction (towards the PSTN). Because lower signal strength will yield less reflected energy, this should clear the problem.
Additionally, you can adjust the receive level, so any reflected audio gets reduced even further. Remember to make small adjustments at a time. Too much attenuation of the signal will make the audio impossible to hear on both sides.
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Alternatively, you can contact the carrier and request to have the lines checked. On a typical T1/PRI circuit in North America, the input signal should be -15 dB. If the signal level is much higher (-5 dB, for example), echo likely will result.
Keeping an Echo Log
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You should keep a log of all calls that experience echo.
Record the time of the problem, the source phone number, and the number called. Gateways have a fixed time of 16 ms of echo cancellation.
If the delay in the reflected audio is longer than this, the echo canceller cannot work properly. This issue should not exist for local calls, and long-distance calls should have external echo cancellers built in to the network at the Central Office. This fact provides one reason why you should note the external phone number of a call that experiences echo.
Checking Your Loads
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Verify your gateway and phone loads. Check www.cisco.com for the latest software loads, new patches, or release notes that may relate to the problem.
One-Way Audio or No Audio
Symptom
When a phone call is established from an IP station through a Cisco IOS voice gateway/router, only one of the parties receives audio (one-way communication).
When a toll-bypass call is established between two Cisco gateways, only one of the parties receives audio (one-way communication).
Possible Cause
An improperly configured Cisco IOS gateway, a firewall, or a routing or default gateway problem, among other things, can cause this problem.
Recommended Action
Make Sure IP Routing is Enabled on Cisco IOS Gateway/Routers
Some Cisco IOS gateways, such as the VG200, have IP routing disabled by default. This will lead to one-way voice problems.
Note |
Before going any further, make sure that your router has IP routing enabled (that is, does not have the global configuration command no ip routing). |
To enable IP routing, enter the following global configuration command in your Cisco IOS gateway:
voice-ios-gwy(config)#ip routing
Check Basic IP Routing
Ensure that basic IP access should always gets checked first. Because RTP streams have no connections (transported over UDP), traffic may travel successfully in one direction but get lost in the opposite direction.
Check the following conditions:
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Default gateways configured at the end stations
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IP routes on the default gateways, mentioned above, leading to the destination networks
Note |
The following list explains how to verify the default router/gateway configuration on various Cisco Unified IP Phones: |
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Cisco Unified IP Phone model 7960/40—Press Settings button, select option 3, scroll down until the Default Router field shows up.
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For Cisco DT24+ Gateways, check the DHCP Scope and make sure that a Default Gateway (003 router) option exists in the scope. The 003 router parameter populates the Default Gateway field in the devices and PCs. Scope option 3 should have the IP address of the router interface that will be doing routing for the gateway. |
Bind the H.323 Signaling to a Specific IP Address on Cisco IOS Gateway/Routers
When the Cisco IOS gateway has multiple active IP interfaces, some of the H.323 signaling may use one IP address for course, and other parts of it may reference a different source addresses. This can generate various kinds of problems, including being one-way audio.
To avoid the problem, the H.323 signaling can be bound to a specific source address, which can belong to a physical or virtual interface (loopback). The command syntax to use under the interface configuration mode follows:
h323-gateway voip bind srcaddr<ip address>. Configure this command under the interface with the IP address to which the Unified Communications Manager points.
Configuring H.323 Support for Virtual Interfaces documents this command, which was introduced in Cisco IOS Release 12.1.2T.
Note |
A bug exists in version 12.2(6) where this solution can actually cause a one-way audio problem. For more information, refer to bug ID CSCdw69681 (registered customers only) in Cisco Software Bug Toolkit (registered customers only). |
Check that Answer Supervision Is Being Sent and Received Correctly from the Telco or Switch
In an implementation that has a Cisco IOS gateway connected to a Telco or switch, verify that answer supervision gets sent correctly when the called device behind the telco or switch answers the call. Failure to receive the answer supervision will cause the Cisco IOS gateway not to cut through (open) the audio path in a forward direction which causes one-way voice. A workaround involves the need to configure voice rtp send-recv on.
Cut-through Two-Way Audio Early Using voice rtp send-recv on Cisco IOS Gateway/Routers
The voice path gets established in the backward direction as soon as the RTP stream is started. The forward audio path will not be cut through until the Cisco IOS gateway receives a Connect message from the remote end.
In some cases you need to establish a two-way audio path as soon as the RTP channel is opened—before the connect message is received. To achieve this, use the voice rtp send-recv global configuration command.
Check cRTP Settings on a Link-by-Link Basis on Cisco IOS Gateway/Routers
This issue applies to scenarios, such as toll-bypass, where more than one Cisco IOS router/gateway is involved in the voice path and Compressed RTP (cRTP) is used. cRTP, or RTP Header Compression, designates a method for making the VoIP packet headers smaller to regain bandwidth. cRTP takes the 40-byte IP/UDP/RTP header on a VoIP packet and compresses it to 2-4 bytes per packet, yielding approximately 12Kb of bandwidth for a G.729 encoded call with cRTP.
cRTP occurs on a hop-by-hop basis with decompression and recompression on every hop. Because each packet header needs to be examined for routing, enable cRTP on both sides of an IP link.
Also verify that cRTP is working as expected on both ends of the link. Cisco IOS levels vary in terms of switching paths and concurrent cRTP support.
In summary, the history follows:
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Until Cisco IOS Software Release 12.0.5T, cRTP gets process-switched.
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Cisco IOS Software Release 12.0.7T, fast- and Cisco express forwarding (CEF)-switching support for cRTP, which introduced and continue in 12.1.1T.
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In Cisco IOS Software Release 12.1.2T, introduced algorithmic performance improvements.
If you are running cRTP on Cisco IOS platforms (IOS Release 12.1), verify that bug CSCds08210 (registered customers only) (VoIP and FAX not working with RTP header compression ON) does not affect your IOS version.
Verify Minimum Software Level for NAT on Cisco IOS Gateway/Routers
If you are using Network Address Translation (NAT), you must meet the minimum software level requirements. Earlier versions of NAT do not support skinny protocol translation and will lead to one-way voice issues.
The minimum software levels that are required for using NAT and skinny simultaneously specify Cisco IOS® Software 12.1(5)T for IOS gateways to support skinny and H.323v2 with NAT.
Note |
If your Unified Communications Manager is using a TCP port for skinny signaling that differs from the default 2000, you need to adjust the NAT router with the ip nat service skinny tcp port<number> global configuration command. |
The minimum software level that is required for using NAT and skinny simultaneously on a PIX firewall specifies 6.0.
Note |
These levels of software do not necessarily support all the RAS messages necessary for full gatekeeper support. Gatekeeper support occurs outside the scope of this document. |
Disable voice-fastpath on AS5350 and AS5400
The Cisco IOS command voice-fastpath enable gets a hidden global configuration command for the AS5350 and AS5400, which is enabled by default. To disable it, use the no voice-fastpath enable global configuration command.
When enabled, this command caches the IP address and UDP port number information for the logical channel that is opened for a specific call and prevents the RTP stream from getting to the application layer, but rather forwards the packets at a lower layer. This helps marginally reduce CPU utilization in high-call-volume scenarios.
When supplementary services, such as hold or transfer are used, the voice-fastpath command causes the router to stream the audio to the cached IP address and UDP port, disregarding the new logical channel information that was generated after a call on hold was resumed or a transfer was completed. To avoid this problem, traffic should go to the application layer constantly, so redefinition of the logical channel gets taken into account, and audio gets streamed to the new IP address/UDP port pair. That explains why you should disable voice-fastpath to support supplementary services.
Configure the VPN IP Address with SoftPhone
Cisco IP SoftPhone offers the ability to make a PC work like a Cisco Unified IP Phone model 7900 Series phone. Remote users who connect back to their company network through VPN need to configure some additional settings to avoid a one-way voice problem.
The solution requires you to configure the VPN IP address, instead of the IP address of the network adapter under the Network Audio Settings.
Verification
A useful command to verify packet flow specifies debug cch323 rtp. This command displays packets that the router transmits (X) and receives (R). An uppercase character indicates successful transmission/reception whereas a lowercase character indicates a dropped packet. See the following example:
voice-ios-gwy#debug cch323 rtp RTP packet tracing is enabled voice-ios-gwy# voice-ios-gwy# voice-ios-gwy# voice-ios-gwy# voice-ios-gwy# !--- This is an unanswered outgoing call. !--- Notice that voice path only cuts through in forward !--- direction and that packets are dropped. Indeed, !--- received packets are traffic from the IP phone to the PSTN !--- phone. These will be dropped until the call is answered. Mar 3 23:46:23.690: ****** cut through in FORWARD direction ***** XXXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXr XrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXr XrXrXXrrrrrrrrrrrrrrrr voice-ios-gwy# voice-ios-gwy# !--- This is an example of an answered call: voice-ios-gwy# voice-ios-gwy# *Mar 3 23:53:26.570: ****** cut through in FORWARD direction ***** XXXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXr XrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXrXr XXrrrrrXrXrXrXrXrXrXrXrXrXrXrXrrXXrrXrXrXrXrXrXXXXXXXXXXXXXXXXrXXXXXXXXrXrXrXXrrXr XrXrXrXrXrXrXrXrXXrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr !-- At this point the remote end picks up the phone. *Mar 3 23:53:30.378: ****** cut through in BOTH direction ***** XRXRXRXRXRXRXRXRXXRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRXRXRXRXRXRXRXR XRXRXRXRXRXRXRXRXRXRXRXRXRXRXXRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRXRXRXRXRXR XXRRXRXRXXRRXRXRXRXRXXRXRXRXRXRXRRXRXXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXR XRXRXRXRXRXRXRXRXRXRXRXRXRXRXRRRRRRRRRRRRRRRRRRRRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXR XRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXXRRRRRRRRRRRRRRRRRRRRRRRRRRRRXRXRXRXRXRXRXRXRXRXR XRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRR RRRRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRRXXRXRXRXRXRXRRXRXRXRXRXRXRXRXR XRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXXRRRRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXR XXRRRRRRRRRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXR XRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXRXR XRXRXRXRXRXRXRXRXRXRXRXRXXRRRXR !-- End of conversation.