Voice transmission is analogical, whereas the data network is digital. The process to sample analogical waves into digital information is made by an encoder-decoder (CODEC). There are many standards to sample an analogical voice signal into a digital one. The process is often quite complex. Most of the conversions use pulse code modulation (PCM) or variations
In addition, the CODEC zip the sequence of data, and sometimes provides echo cancellation. The compression of the waveform can save bandwidth. This is especially interesting in low speed connections so you can have more VoIP connections at the same time. Another way to save bandwidth is using the silence suppression. The goal is not to send packages when there is no voice in the conversations.

Softphone Avec Codec G729 Bow Wow Ft Omarion Girlfriend Mp3 Download Free Prs Serial Number Handwritten Signature Ottimo Perfect Cut Keygen Serial Biochemistry And. Test codecs Test with all the codecs g711u, g729 and GSM. Sometimes the issues with the audio can be related with the codecs in use, either because the codec we are using is consuming too much bandwidth for our connection, there is also a chance the device we are using is not supporting this codec very well or it works better with a different one. If, 3cx client can use codec G729, I number of calls to use G729 depends on the license that you have of 3cx phone, 3cx only uses G729 for the external extensions. Next I number of G729 channels that you can use following your license.the mini Edition includes two G729 channels, the Small Business Edition includes 4 sim G729 calls, The Pro.

Next is a table with the most known codecs in use:
- Bit Rate - The rate at which bits are transmitted over a communication path. Normally expressed in Kilobits per second (Kbps)
- Sampling Rate - the number of samples taken per second when digitizing sound. The quality of the digital reproduction improves as the number of samples taken per second increases.
- Frame size - The time between packets sent
- MOS - (Mean Opinion Score). It is a subjective measure of sound quality from 1 to 5.
In order to understand better the codec process and the parameters expressed in the table we recommended to read the section of G.711 codec process where it is possible to learned how it works the G.711 codec.

Number	Standard by	Description	Bit rate (kb/s)	Sampling rate (kHz)	Frame size (ms)	Remarks
G.711 *	ITU-T	Pulse code modulation (PCM)	64	8	Sampling	U-law (US, Japan) and A-law (Europe) companding	4.1
G.711.1	ITU-T	Pulse code modulation (PCM)	80-96 Kbps	8	Sampling	Improvement og G.711 to provide an audio bandwidth of 50 Hz to 7 kHz More info
G.721	ITU-T	Adaptive differential pulse code modulation (ADPCM)	32	8	Sampling	Now described in G.726; obsolete.
G.722	ITU-T	7 kHz audio-coding within 64 kbit/s	64	16	Sampling	Subband-codec that divides 16 kHz band into two subbands, each coded using ADPCM
G.722.1	ITU-T	Coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss	24/32	16	20
G.722.2 AMR-WB	ITU-T	Adaptive Multi-Rate Wideband Codec (AMR-WB)	23.85/ 23.05/ 19.85/ 18.25/ 15.85/ 14.25/ 12.65/ 8.85/ 6.6	16	20	is mainly used for speech compression in the 3rd generation mobile telephony. More info
G.723	ITU-T	Extensions of Recommendation G.721 adaptive differential pulse code modulation to 24 and 40 kbit/s for digital circuit multiplication equipment application	24/40	8	Sampling	Superceded by G.726; obsolete. This is a completely different codec than G.723.1
G.723.1	ITU-T	Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s	5.6/6.3	8	30	Part of H.324 video conferencing. It encodes speech or other audio signals in frames using linear predictive analysis-by-synthesis coding. The excitation signal for the high rate coder is Multipulse Maximum Likelihood Quantization (MP-MLQ) and for the low rate coder is Algebraic-Code-Excited Linear-Prediction (ACELP).
G.726	ITU-T	40, 32, 24, 16 kbit/s adaptive differential pulse code modulation (ADPCM)	16/24/32/40	8	Sampling	ADPCM; replaces G.721 and G.723.	3.85
G.727	ITU-T	5-, 4-, 3- and 2-bit/sample embedded adaptive differential pulse code modulation (ADPCM)	var.	Sampling	ADPCM. Related to G.726
G.728	ITU-T	Coding of speech at 16 kbit/s using low-delay code excited linear prediction	16	8	2.5	CELP.	3.61
G.729 **	ITU-T	Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)	8	8	10	Low delay (15 ms)
G.729.1	ITU-T	Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)	8/12/14/16/ 18/20/22/24/ 26/28/30/32	8	10	Improvement og G.711 to provide an audio bandwidth of 50 Hz to 7 kHz More info
GSM 06.10	ETSI	Regular­Pulse Excitation Long­Term Predictor (RPE-LTP)	13	8	22.5	Used for GSM cellular telephony.
LPC10	USA Government	Linear-predictive codec	2.4	8	22.5	10 coefficients.
Speex	8, 16, 32	2.15-24.6 (NB) 4-44.2 (WB)	30 ( NB ) 34 ( WB )
iLBC	8	13.3	30
DoD CELP	American Department of Defense (DoD) USA Government	4.8	30
EVRC	3GPP2	Enhanced Variable Rate CODEC	9.6/4.8/1.2	8	20	Se usa en redes CDMA
DVI	Interactive Multimedia Association (IMA)	DVI4 uses an adaptive delta pulse code modulation (ADPCM)	32	Variable	Sampling
L16	Uncompressed audio data samples	128	Variable	Sampling
SILK	Skype	From 6 to 40	Variable	20	Harmony codec is related with SILK

Softphone With G729 Codec Free Download

* G711 has two versions called U-law (US, Japan) and A-law (Europe) . U-law is in relation with the T1 standard used in North America and Japan. The A-law is relation with the E1 standard used in the rest of the world. The difference is the method to sample the analog signal. In both schemes, the signal is not sampled linearly, but in a logarithmic way. For more information about the differences you could visit G.711 A Law versus u Law.
** There are different versions of g729 codec that it is interesting to explain because this codec is very used nowadays.
G729: original codec
G729A or A annex: it is a simplification of G729 and it is compatible with G729. He is less complex but it has less quality.
G729B or B annex: G729 with silence suppression and not compatible with the previous ones
G729AB: g729A with silence suppression and only compatible with G729B.
Besides, every version of G729 have 8Kbps of bitrate but there are versions with 6.4 kbps (D annex) and 11.4 Kbps (E annex).

From VoIP.ms Wiki

This is the published version, approved on 6 November 2020. Accuracy Accurate Depth Moderate Readability Good

Article en Français	Artículo en Español
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There are different factors that could potentially affect the quality of your calls. There are several types of sound issues and these can be related to different causes. We will try to mention here some suggestions, so we can identify which type of issue we are experiencing and what things we need to check to start diagnosing on our own.

Contents 4Choppy/Robotic voice

Reboot your device

Even if you can browse the internet without any issues and if you think the Internet is working fine, it is possible that something in the network is affecting the calls. The first thing that always needs to be tested is to reboot the ATA device and Router, this way we refresh the connection.

Choose a server

A good recommendation is to send a ping to all the available servers, this way you can verify the latency and pick the best option available for your network. (This is just a slight introduction, please refer to our article Choosing Server for more information.

Softphone test

To rule out if your ATA Device or PBX is the source of the issue, you can do a test with a simple software that can be used for the same purpose (make calls).

How to test using Softphones?

• Create a sub account, this way you do not have to alter the settings on your ATA device for the moment.

• Register the softphone using the sub account credentials and make a call, if the issue is the same, the problem can be in our network, if not, then we can start pointing to your device.

• ZoIPer and X-Lite are recommended by the VoIP.ms staff as they are easy to configure. We also recommend Jitsi, with this softphone you can call the Echo test (4443), put the call on pause and verify the jitter and packet loss values. This can be more reliable then sending a ping because the ping may not be prioritized.

Choppy/Robotic voice

Network traffic

One of the main reasons sound issues may occur is based on the traffic or congestion on the network. First thing to try is check if the issue can be duplicated by making an internal call with the provider, for example using an Echo test application (by dialing 4443) or a voicemail.

Some symptoms that can be present because of the lack of bandwidth available:

• Audio cutting in and out (choppy).

• Voice sounding robotic, like if you were talking under water.

• Audio slowing down or speeding up intermittently during the call.

To test if the bandwidth is affecting our calls:

• Disconnect all the devices from the network

• Disable wireless to make sure no one else is using your internet.

• If your router has QoS, disable it.

• If you were using software to download stuff from Internet (e.g. Torrents) wait a few minutes for this traffic to subside.

After following all these suggestions, use a single device and try to make a call If the audio quality is fine, you are probably dealing with lack of bandwidth, and in this case the use of QoS is recommended ( be certain the set up is well done).

Test codecs

Test with all the codecs g711u, g729 and GSM. Sometimes the issues with the audio can be related with the codecs in use, either because the codec we are using is consuming too much bandwidth for our connection, there is also a chance the device we are using is not supporting this codec very well or it works better with a different one. In any case, this test can also help in the diagnostic.

Check in your Account or sub account settings, which codec you are allowing, you can test allowing one by one, until you get the best result. If using codecs such as G.711 you may try with a lower bitrate codec such as G729a or GSM (if they are supported by your device/software/system).

Check your ISP

After following these suggestions and you still experience sound issues, you may consider contacting your ISP (Internet provider) just to confirm the issue is not related with them.

Common ISP-related issues include:

• Shared connections. Your ISP advertises 'up to' some particular speed, but guarantees absolutely nothing as the minimum. Odds are, those megabits per second are shared with other subscribers of the same ISP so when they go online, your connection slows down unpredictably. Your service works very well... except in peak hours when everyone is online, then suddenly 'you're breaking up' or experiencing connection problems.
• Asymmetric connections. You have 'up to' a few megabits of what your ISP insists is 'blazingly fast' speed for downloads, but they forgot to mention that your upload speed is a tenth that figure or worse. Pick up the 'phone and you hear the other party easily, but they say 'you're breaking up' and ask you to constantly repeat things.
• Throttling. Sometimes, an ISP knows they've oversold their bandwidth and, if all of their subscribers go online at once, they will have problems handling large downloads. As a means of damage control, they start playing favorites. Hopefully, they put time-sensitive traffic first (such as voip.ms, which only needs about 80 kilobits per second - and not megabits - for each call) and less time-sensitive downloads last, but there is a risk: some of these schemes allocate bandwidth in a sporadic manner, so that a connection is fast for one second and breaking up the next. That's not good for Internet telephone calls of any kind.

If you have applications which purport to send voice free to other users of the same Internet app (Netmeeting, webcam, Skype...) try an Internet-to-Internet call during the time periods when the problems are at their worst. If your webcam audio breaks up too, the problem might not be VoIP but your ISP. Running Internet 'speed test' applications to see if the results are varying widely between attempts may also be very telling.

This isn't a guarantee that your ISP will own up to the issues, let alone fix them, but if your ISP is the problem no voice apps will work.

Tones during calls

Another issue related with the quality during your calls, is when you can hear beep tones during a call, like if someone is pressing a button on the phone or trying to dial. This is usually known as 'talk-off' and the device is interpreting the voice as a DTMF digit.

Suggestions to follow:

• Upgrade the firmware in your device, sometimes these bugs are fixed in recent versions.

• Change your DTMF Tx Method to InBand (you have to change this setting in your device and in your account or sub account settings). Test if the DTMF tones are working fine, dial 4747 for this test.

• If Inband doesn't work for you, test with DTMF Process INFO and DTMF Process AVT to No, if the options are available in the device.

• Another alternative you can do: DTMF Tx Method: AVT, DTMF Tx Mode: Strict, DTMF TX Strict Hold Off time: 70.

Echo during calls

We have different factors that can cause Echo during the calls, we will review some suggestions to work with:

• Check the volume on the phone is not too loud, it is possible the phone is causing the issue.

• Make a call dialing 4443 for echo test and see if you can reproduce the same situation with this test.

• Again, check the firmware on the device, usually this can help to reduce the echo if you do not have the latest firmware.

• The default gain on some devices, is typically too high and can cause echo. For instance on Cisco PAP devices, you can adjust the FXS Port Input Gain and FXS Port Output Gain, one at a time, in increments of three. You can test using -1 and -11.

Note: Input Gain = how you sound to the other party. Output Gain = how the other party sounds to you.

• If the above does not solve your issue, and you have a Linksys device, verify that Echo Canc Enable, Echo Canc Adapt Enable, and Echo Supp Enable are set to Yes. (These are default settings.)

• If you use laptop (integrated mic/speakers), echo can be caused by microphone catching noise from speakers. Try lowering MIC Input sensitivity. Using a headset instead of the microphone and speakers will greatly reduce the amount of noise heard by the other party.

One-Way Audio

You can hear the other party but they can not hear you, and vice-versa. When a situation like this is present, it is know as 'one-way audio' and usually it is related with the NAT. The primary cause for one way audio is the NAT enabled device hiding the topology of the customers network. Many legacy devices do not have a built in ALG (Application Layer Gateway), which changes the headers of the VoIP packets (either SIP or MGCP) to allow the customer to preserve their private network topology and allow them to use VoIP service.One-way audio is caused when one side of the RTP stream is not setup or terminated correctly. RTP is the UDP media stream that carries the audio of a phone call on VoIP. Let's try with the following suggestions:

• From the account or sub account settings, select always NAT=Yes (this is the option recommended by VoIP.ms).

• Try using each codec in a separate way, starting with G711u codec only, from the customer portal > Main menu > Account settings > Advanced tab > allowed codecs.

• Only as a test, place the device in DMZ to test if the issue is related with the NAT. do not leave the device in DMZ after finishing troubleshooting.

• Is your router appropriate for VoIP? If you have a router and a modem, try to bypass the router to verify if the issue gets duplicated.

• If your router includes a SIP ALG and/or SPI Firewall setting please ensure that it is disabled. That setting is common in D-Link and Netgear routers. If this does not help make sure you are using the most recent firmware version for your device.

• You can see what port is being used for audio by looking at the UDP port 5060 traffic. The RTP traffic will typically be in the UDP port range 10000-20000.

• Try to use a Softphones like Zoiper and X-lite to try and duplicate the issue, there are free version.

Softphone With G729 Codec

• Reset your device. In some cases this action resolves the issue.

• If you have an ATA device (Linksys) you can work on the following settings:

Under SIP page.

• RTP Packet Size: 0.020
• G729a Codec Name: G729
• G729b Codec Name: G729

Under the Line page.

• NAT Mapping Enable: Yes
• NAT Keep Alive Enable: Yes
• Preferred Codec: G711u
• Use Pref Codec Only: No

Contact your provider

Could it be the case my quality issue resides on the VoIP provider? Yes, it is possible. Some things we can check and specify to provider when opening the ticket are:

• Are the sound issues present only with incoming calls or only with the outgoing calls, or both? Are the sound issues present while dialing 4443 to reach echo test?

• If issues are present only when calling certain areas or specific countries.(please provide the number(s) when opening the ticket)

• If the issue are happening only with the incoming calls, then please route your DID to echo test, and call it from an external provider (preferably landline) and check if the issues appear when doing this test.

• If you are unable to receive incoming calls on one DID, do the DID's call attempts appear on the Call Detail Records? If not, the DID may be broken at an upstream provider level - especially if other-city DIDs on the same account and same telephone handset are working normally.

• For Canada, if the issue is present with outgoing calls only, test using the different options for Routing (value or premium). If the sound issue happens with one route and not with the other, then you need to contact the provider.

Portions of this article have been taken from 'How to Troubleshoot Poor VoIP Audio Quality' by Mango. Used with permission.