Install Ilbc Codec Asterisk Phone

How To Install Ilbc Codec In Asterisk Voip. 7/30/2017 0 Comments. If you don’t have Viber on your phone and try to install the service on your PC. The Internet Low Bitrate Codec (iLBC). Install Ilbc Codec Asterisk Voip Requirements. VoIP phone systems, and Asterisk telephony software.

UPDATE: This may not work any more because Google said that they will cut off the Google Talk XMPP on 5/15/2014, which PBXES relies on. Mp3. This guide is not about how to make free calls. It's about how to make free calls using iLBC codec so that you can make reliable calls over 3g connection. If you only want the instructions, then skip to part V.

(edited) I have changed my VoIP set up to PIAF on AWS EC2. See my signature. BACKGROUND There are several known methods to make free VoIP (internet) calls over 3g. I'll briefly discuss each to explain why there's no perfect solution. I'm not claiming my solution is perfect, either.

You have to try them out and find out yourself what works for you. Groove IP app codec: PCMU (aka G711u, uncompressed signal used by PSTN and Google Voice, only good for fast/reliable wifi connection) pros: simple setup, works well on wifi. If you use VoIP only on wifi, this is for you. The voice quality is terrible on mobile data (3g) connection.

Talkatone app codec: speex for standard quality pros: easy setup, good quality on 3g using speex. Cons: doesn't integrate well with android 3. Vonage mobile app codec: G729(?) when making regular calls. ISAC when calling a Vonage member.

Pros: G729 provides an excellent quality even on 3g. Cons: can't receive calls and doesn't integrate with Android. You need to input 10 digit number to make a call, unless the number is in your contacts. Google Voice/pbxes/sipdroid () codec: lots of options based on connection type pros: excellent battery life (), If you choose speex codec for 3g, then call quality is pretty good on 3g.

Cons: must unlock the screen before answering the call. Doesn't support better codecs such as iLBC/G729. Simonics GV gateway + sip client of your choice codec: PCMU pros: simpler setup than pbxes, some ppl claim it's more reliable. Cons: limited codec support.

No multiple registrations 6. Setting up your own asterisk server () pros: everything is configurable cons: requires a 24/7 running server I have been primarily using #4 setup but wasn't satisfied with the call quality on 3g so I was about to do #6. But I found a way to tweak #4 to use iLBC codec and greatly improved the call quality on 3g. CODEC COMPARISONS The call quality basically depends on (1) the codec that your phone uses and (2) the network throughput. Unfortunately, there's no perfect solution because of trade-offs.

PCMU (64kbps) is the uncompressed signal that is transmitted at PSTN. So it will give you the best quality as long as your network throughput supports it. This is the preferred codec for wifi connection. However, 3g network cannot consistently sustain 64kbps and will suffer from packet loss resulting in jitter. This explains why Groove IP works best on wifi but not over 3g.

At the other end of the spectrum is GSM codec (8kbps). It's designed in 1990 for wireless communication over 2g. It's the most compressed signal, which is used by your regular GSM providers (AT&T, Tmo). These wireless providers allocate sufficient capacity on voice calls so your voice calls will suffer little jitter. However, the call quality is inferior to PCMU codec.

There are many codecs in between. For example, Sipdroid and talktatone can use speex, which requires low bandwidth. These apps can send the same signal twice to compensate for packet loss. The 2 codecs that haven't been used widely in commercial apps are G729 and iLBC. These 2 codecs require low bandwidth (DID forwarding.

Input your in the box and save. AFAIK, callcentric doesn't charge call forwarding to a SIP address (our method). However, it does to a regular phone number. Alternatively, #7b.

Have pbxes intercept the calls to callcentric. Log in to pbxes.org. Click add trunk, select SIP trunk and input callcentric credentials. (user name starts with your callcentric 1777 number.) Select 'audio bypass' to pass-through callcentric's iLBC signal to the phone. See the attachment for what my setup looks like. Here's a tutorial on adding trunks: #8. Add extension(s) in pbxes.

Log in pbxes.org. Click extensions. If you followed step #4, you should see extension 'Sipdroid-200.'

You can add new ones or modify extension 200. Change the password to something that you can easily remember. This will be the password you will use to register your phone to pbxes in the next step. Finally, select 'audio bypass.'

Install nightly version csipsimple app from. The play store version crashed with iLBC. First, add the pbxes account. If you are new to VoIP, use the wizard for pbxes. User name should be yourgoogleID-200, where 200 is the extension you used in #8.

The password is what you changed to in #8. (I recommended you change the password in #8 so that you don't use your gmail password everywhere.) Do not add callcentric account because having 2 registrations drains battery faster. If you followed the steps above, you will still be able to receive callcentric calls from your pbxes account. I prefer csipsimple because it is has lots of options to configure. But it can be too overwhelming to some people. If you can't get it to work, then try other sip clients such as media5-fone. Let's change csipsimple for better call quality.

First, in settings/media/codecs, select iLBC as the only codec for 3g connection. If you have a strong wifi connection, then select PCMU for wifi connection. (optional) If you experience looping when making calls, set up filters appropriately. I set up mine such that all calls will be made by pbxes except for 911.

Google Voice (from webpage) Add callcentric DID number (not the 1777 number). All the GV forwarded calls to callcentric will be forwarded to pbxes and ring csipsimple on your phone. Uncheck Google chat so that pbxes only receives callcentric forwarded calls. FYI, GV doesn't support iLBC codec natively. (optional) uncheck Tmo mobile number, if you are completely satisfied with this setup.

If GV is forwarding to your cell, then your cell will be ringing twice and you may accidentally answer the cell phone using your minutes. (optional) Test in/out calls.

I use these numbers to test the audio quality. *43 (pbxes echo test) (909) 390-0003 (PSTN echo test) (408) 647-4636 (record/playback) Note that during a call in csipsimple, you can click the settings button in the lower right corner to change the in/out volumes and enable/disable echo cancellation. You should also test incoming call quality using Google Voice call back. This is where I had the most troubleshooting. My incoming call quality was awful until I found callcentric trick.

I have tested this setup in many different scenarios including in a car running at 45 mph and very happy with the results. Of course, YMMV depending on how good your 3g connection is. LIMITATIONS What are the drawbacks? Latency (audio delay). ILBC codec makes CPU interpolate to make up for loss packets. This is why the call quality is better than other codecs over 3g. However, the disadvantage is it increases audio lag.

I'm experiencing a 0.5s lag one way on 3g. This is not too bad considering a cell>GV>cell call has a similar latency (, couldn't find a more recent test result) 2. Once a caller calls your GV number, it takes 5-6 seconds for your phone to ring. This is a limitation of PSTN and GV forwarding not VOIP.

On the other hand, if a caller calls my cell number directly, it rings in 4-5 seconds. So all the call routing through caller>Google Voice>callcentric>pbxes>phone will add 1 extra second for your phone to ring compared with caller>phone.

Pbxes with gtalk trunk will make your Google chat status show as 'online and available' all the time. Your friends may think you are online when you are actually not. Another side effect is you won't be able to answer GV calls from PC gmail any more. I haven't found any workaround to fix this other than creating a new google voice account. Nexus 4 VoIP issues. It has been reported that several hardware mic functions (echo/noise cancellation) are disabled for N4 VoIP. OTOH, echo cancellation is enabled in Galaxy Nexus.

You can mitigate the issue with adjusting the media settings in sip client. But if you are in a noisy area, the mic is going to pick up all the noise. Consider using the headset.

TIPS/TRICKS 1. If your family/friends know how to internet call, then it's best to give out your SIP address;. Then the call will go out fast and the phone quality will be the best because you are not using PSTN at all. If both caller and recipient are using the same pbxes server, then pbxes is the only node between the two. For example, when my wife with the same setup calls me @es, my phone rings in less than 2 seconds and the call quality is the best.

Create multiple extensions (300, 400, etc) in pbxes. I have 3 devices registered to pbxes simultaneously: my cell phone, my tablet and the ATA for my home POTS phone.

When I receive a GV call, all 3 devices ring simultaneously. And I can call any other extension independently by entering its 3 digit extension number. Try other sip clients I suggested csipsimple because it's open source and free despite being an unstable nightly version. You should try out other apps because they may perform better for you. For example, when I use media5-fone (free w/ ads, $5 to remove ads), the audio latency improved but it consumed a little more battery. Battery improvement My Nexus 4 easily lasts 16 hours w/ 3+ hour screen time.

There are several things you can do to improve battery life. Use TCP only in client to communicate with pbxes.org.

If you use csipsimple, you need to change the account settings. First, use the wizard to switch to expert mode. Then you will see 'transport' option in the account settings. Another thing you can do to improve battery life is to experiment with larger keep alive intervals.

Use a different gmail account for your GV number If you are uncomfortable with giving out your gmail password (even the app specific one) to pbxes, then transfer your GV number to another gmail account. The whole process took less than 30 minutes. Once done, you don't need to create a new pbxes account. Just log in pbxes.org, go to Trunk>Gtalk and make the necessary changes. If done properly, you can make calls out again instantly.

If you are using GV app, then you need to add the new account. Delayed GV call forwarding You can set up pbxes such that it rings your cell phone if you don't answer the internet call in X seconds. Troubleshooting 1. This guide is definitely not simple.

Lots of things can go wrong. If you want help, then provide as much information as possible. Posting a screenshot of pbxes call monitor log will be a good start. If csipsimple doesn't work, try other sip clients such as media5-fone. If you can't make a call with media5 either, then you know at least it's not csip's fault.

If you suspect pbxes malfunctioning, change the datacenter (server) from pbxes.org.

Open Source G.729 and G. Hauppauge Wintv 8 Serial Number there. 723.1 for Asterisk/FreeSWITCH Download Open Source G.729 and G.723.1 codec_g729 and codec_g723.1 for Asterisk Asterisk 1.0.x Asterisk 1.2.x Asterisk 1.4.x Trunk, tested SVN 42264 Asterisk 1.6 and later, FreeSWITCH • • • • • • You can browse the Credits • Intel originally offered a sample G.729 and G.723.1 implementation for Linux. • Daniel Pocock converted Intel's code into an Asterisk module. • Arkadi Shishlov and others supplied compiler optimizations, and bug fixes. Introduction This code let's Asterisk use the G.729 and G.723.1 protocols for voice compression when communicating with other devices. The code produces a Asterisk modules, codec_g729.so and codec_g723.so, that you put in your Asterisk modules directory. The code is provided as a patch which will convert Intel's sample application into an Asterisk codec module.

This code only runs on modern Intel processors that are supported by Intel IPP libraries. See for details and license. Why G.729 or G.723.1? People often ask the following questions: • What is a codec? • Which codec should I use? • Which codecs does my phone support?

• Which codecs use the least bandwidth? • Apart from bandwidth, what other issues should I consider when deciding on a codec? • Why should I use the G.729 codec or G.723.1 In VoIP networks, codecs are used to compress regular audio (16 bit signed linear audio, usually sampled at 8000Hz). Codecs are usually `lossy'.

This means that the output data does not have to be perfectly identical to the source data, it just has to sound the same when converted to sound. If your VoIP network is on an office LAN and the signal doesn't ever traverse a WAN connection (internet, VPN, DSL, etc), then compression isn't critical. If your VoIP signals may need to traverse a WAN, then you need to compress the signal as much as possible.

This allows you to fit more simultaneous phone calls into a single WAN connection. Compression also creates smaller packets. Smaller packets means less audible delay and lower risk of packet loss. G.729 is often preferred as a codec for WAN applications because of the following: • Many devices offer only 1 or 2 low bit rate codecs, usually G.729 and one other or just G.729.

If you have bought phones that only support G.729, then you have little choice. • Some gateway providers will only allow you to talk to their gateway with G.729. • A good G.729 implementation uses less bandwidth and less CPU power than other low bit rate codecs such as iLBC. G.729 uses 8kbps, iLBC uses 13kbps.

• Some people have observed their CPU performing up to 50% better when doing G.729 compression compared to iLBC. • Few phones implement iLBC (one such phone is Budgetone 101 and 102). Many others - Cisco 7940, Snom, Swissvoice - only offer G.729 • Most phones offer G.711 (ulaw/alaw) as well - that is actually 64kbps, eight times the bandwidth required by G.729. It is only for use on LANs.

G.723.1 is used for similar reasons to those just listed, but gives the benefit of using even less bandwidth but with a more noticable degradation of sound quality. Why NOT G.729? There are some reasons you might not want or need to use G.729. • You don't want to pay the license fees or use the codec without the permission of the patent holder. • You don't want to use extra CPU cycles to perform compression, possibly because CPU power is more expensive than extra bandwidth where you live. • Your VoIP network is all running on ethernet.

See also, mirrors Legal Stuff - Important, please read To use G.729 or G.723.1 you may need to pay a royalty fee. Please see for details. Please note that this code is available for you to download for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their permission before executing the code. To distribute Intel's IPP libraries with a commercial product, you may need to pay a once-off license fee to Intel (currently $US180).

My patches to Intel's code are distributed free under the GPL. Most of the code is just Intel's sample code re-arranged a little bit to work the way Asterisk expects. Therefore, this work would not have been possible without Intel doing 95% of the work.