Troubleshooting: Voice quality issues

Ultimately, the quality of Voice over IP calls depends on the quality of the network components and end devices involved. These have a significant influence on interference factors such as latency, jitter, and packet loss. Private users and businesses should consider this when selecting their network components.



Possible problems that may occur:

  • Disconnection «Call interrupted»

  • Metallic language «robot voice»

  • Echo «own voice audible»

  • Interruptions in the audio «dropouts, stuttering»

  • No audio «Codec, Firewall, NAT Traverse»

  • No signaling «Transport Protocol TCP/UDP»

The problem can occur on both sides—yours and/or the other party's. If you experience the issue on multiple calls to/from different parties, it's likely on your side.

peoplefone GmbH

Cody Blaylock

Cody Blaylock

Teamleiter Service

Content Blocks

Problems with voice quality

The best practice is to perform PING tests, especially when there are voice quality issues. Make sure you run the tests on the same network as your VoIP device/client.

You will learn how much packet loss and response time your network requires for our network.



Packet loss should be 0% and response time should be less than 30 ms.



Also run a trace route to find out how many routes you take to reach our network.

Perform a ping/trace route

What does the ping do?

  • Resolves the IP of a domain

  • Checks if the other side is reachable

  • Displays connection quality (response time and packet loss)

PING sips.peoplefone.ch (95.128.80.3): 56 data bytes

64 bytes from 95.128.80.3: icmp_seq=0 ttl=53 time=9.332 ms

64 bytes from 95.128.80.3: icmp_seq=1 ttl=53 time=8.895 ms

64 bytes from 95.128.80.3: icmp_seq=2 ttl=53 time=8.827 ms

64 bytes from 95.128.80.3: icmp_seq=3 ttl=53 time=9.268 ms

64 bytes from 95.128.80.3: icmp_seq=4 ttl=53 time=9.038 ms

64 bytes from 95.128.80.3: icmp_seq=5 ttl=53 time=8.946 ms

64 bytes from 95.128.80.3: icmp_seq=6 ttl=53 time=9.310 ms

64 bytes from 95.128.80.3: icmp_seq=7 ttl=53 time=9.040 ms



--- sips.peoplefone.ch ping statistics ---

8 packets transmitted, 8 packets received, 0.0% packet loss

round-trip min/avg/max/stddev = 8.827/9.082/9.332/0.184 ms

Perform a trace route

What does trace routing do?

  • Resolves the IP of a domain

  • Shows the routes between you and the destination

  • Shows the time each hop took

traceroute to sips.peoplefone.ch (95.128.80.3), 64 hops max, 52 byte packets

1 192.168.43.1 (192.168.43.1) 6,455 ms 5,002 ms 7,254 ms

2 192,168,43,127 (192,168,43,127) 219,503 ms 356,936 ms 46,440 ms

3 192.168.43.77 (192.168.43.77) 19,922 ms 27,107 ms 27,237 ms

4 1787.eth-trunk20.zhbmb00p-cgn002.bluewin.ch (213.3.229.78) 31,624 ms 23,541 ms 21,340 ms

5 ***

6 213.3.229.6 (213.3.229.6) 34,219 ms 23,231 ms 29,420 ms

7 i79zhb-041-bun1.bb.ip-plus.net (138.187.129.25) 38,026 ms 33,289 ms 28,355 ms

8 193,247,171,142 (193,247,171,142) 40,578 ms 29,154 ms 50,009 ms

9 zrh01-fw-c1-p1.peoplefone.net (185.190.124.9) 29,922 ms 19,576 ms 39,269 ms

10 pbxs.peoplefone.ch (95.128.80.3) 32,976 ms 19,315 ms 36,730 ms

What can you do about it

Set bandwidth for VoIP

Ensure that sufficient bandwidth is always available in the network for telephony. Reserving a minimum bandwidth for VoIP and prioritization functions in the switches such as Class of Service ("CoS"), port prioritization, service prioritization, and IEEE 802.1q/Q support can help with this. Depending on the labeling of the voice packets, Quality of Service ("QoS") mechanisms in the switches then intervene to prioritize voice data over less time-critical data such as emails.



Create VLAN for telephony

A Virtual Local Area Network (VLAN) is a logical subnetwork within a switch or an entire physical network. It can extend across multiple switches. A VLAN divides physical networks into subnetworks by ensuring that VLAN-enabled switches do not forward frames (data packets) to another VLAN, even if the subnetworks may be connected to common switches.



Test the internet speed of the line

To obtain general information about your existing internet connection, we recommend running a speed test at https://www.breitbandmessung.de/test. Any other speed test will work, of course; perhaps you could compare our recommended speed test with that of your internet provider, if available. It's important that there are no packet drops; this would generally indicate a poor connection or other problems.



Test response times (latency times)

By pinging one of our servers, you will receive a response from our proxy server. A roughly consistent response time is very good for communication via Voice Over IP.



peoplefone proxies

Depending on whether you have a standard account or a peoplefone HOSTED account, you'll need to ping the correct proxy. This will give you a response from our infrastructure.

  • sips.peoplefone.de «SIP Trunk / Standard Account»

  • pbxs.peoplefone.de «peoplefone HOSTED»

Windows

Press the Windows and R keys on your keyboard and type «CMD», then press Enter



Enter the command «ping -t sips.peoplefone.de».



Reply from 95.128.80.8: bytes=32 time=9ms TTL=54

Reply from 95.128.80.8: bytes=32 time=9ms TTL=54

Reply from 95.128.80.8: bytes=32 time=9ms TTL=54

Reply from 95.128.80.8: bytes=32 time=9ms TTL=54



If the times differ significantly, usually from around 23ms, this indicates typical irregularities in the network, or the latency times are very high and can cause problems and disruptions in Voice over IP telephony. The lower the value, the faster the response times and therefore the better the performance.



Reply from 95.128.80.8: bytes=32 time=37ms TTL=54

Reply from 95.128.80.8: bytes=32 time=215ms TTL=54

Reply from 95.128.80.8: bytes=32 time=112ms TTL=54

Reply from 95.128.80.8: bytes=32 time=62ms TTL=54



You can cancel the process with CTRL-C or simply close the command window



Apple / MacOS

If you have a MAC, you can start the terminal under «Applications» «Utilities» «Terminal».

Then enter «ping pbxs.peoplefone.de» without brackets.

You can cancel the process with CTRL-C or simply close the command window.

Codecs

Codecs, as the name suggests, are used to "encode" and "decode" audio. This is simply the technical term for translating a signal into another format. In this case, we take the analog audio signal from the microphone and translate it into a digital signal that can be sent over the internet.



And vice versa for the incoming audio.



Both sides negotiate to use one of these codecs to successfully establish a call.

This means that both sides of a call must support at least one of these codecs.



Supported codecs

Peoplefone supports calls with the following codecs:

  • G.711a (also known as «a-law» or «PCMA»)

  • G.711u (also known as «u-law» or «PCMU»)

  • G.722

  • G.729a

G.711a/u are the most commonly used codecs and have a quality comparable to normal landline calls.

G.711a is commonly used in the Europe/Asia-Pacific region, while G.711u is more commonly used in the Americas. While similar, they are not interchangeable, so we recommend enabling both for best compatibility.



G.722 is an HD codec that offers better quality but also requires more bandwidth. As mentioned above, both sides must support this codec.



G.729a is a codec that uses a special compression algorithm to achieve the same quality as G.711a/u, but with less bandwidth requirements.

Due to the algorithm, this is a proprietary codec, meaning it requires a paid license and is therefore often sold as an add-on in otherwise free VoIP software products, while many desk phones have it licensed from the manufacturer and as such it is available for the phones without any additional purchase.

Possible reasons for a busy network

Internet

Here are a few examples that could cause problems with Voice Over IP telephony:

  • No good internet coverage in the region

  • Not the performance promised by the Internet provider «slow and high latency»

  • Not the optimal hardware

  • Insufficient internet service booked



Servers and services

Servers or services that perform certain tasks on the network can therefore also drain the performance of the respective network. In the example of an FTP server where files are copied back and forth, network performance is required for this time. The same applies to a WSUS ("Windows Server Update Service"), which delivers updates to clients over lunch after downloading them; this could potentially lead to performance fluctuations on the network. But access to NAS, backup systems, etc. can also drain the performance of the network. This can also lead to bottlenecks even with a "glass connection." If the hose becomes too small, no more packets can get through.

  • FTP

  • Web server

  • WSUS

  • NAS

  • Backup systems

  • DNS

  • etc.



Hardware

It's not always a server or service that's to blame for poor performance; the clients/PCs downloading an update may also be to blame for degraded network performance. However, it's also possible that a cascading of switches, the cabling, or even a faulty switch port are causing the problem.

  • Headphones and microphone

  • Client Computers «Client Updates»

  • Cascading «Switch1 → Switch2 → Switch3»

  • directional antenna

  • Routers, firewalls, switches

  • Cabling cables «Cat5, Cat6»

  • etc.



Programs / Applications

Programs and applications can also require performance on the network, as the amount of data moving across the network increases over time. Here are just a few possible applications that are performance hogs:

  • Office 365

  • Video / audio streaming

  • Security checks of virus programs

  • Video and image editing

  • CAD

  • Heavy files «large files»

  • etc.

Comments