Iaxclient g729 työt

Looking for someone to build a basic flutter Android app for sip client with dart. This will be used for basic call testing with will will build further options with new screens later. For your ref a brief code is attached.

I am looking for a skilled freelancer who can assist me with a text to voice conversion project. The ideal candidate should have experience working with G729 format. Specific Requirements: - Text to voice conversion in G729, G723 format - Ability to generate multiple language and accent outputs as required - Conversion of approximately 10 to 20 seconds of audio - Text/SMS will be sent from an SMS Switch, it can be sent using SMPP or web API Skills and Experience: - Proficiency in working with G729 format - Familiarity with various languages and accents for voice output - Strong understanding of text to voice conversion techniques - Attention to detail and ability to deliver accurate and high-quality results

I am looking for a skilled developer to create a project that involves converting text from the SMPP protocol into speech. The desired output format for the converted speech should be G729 and G723 Protocol. Specific requirements for this project include: - Support for multiple languages, as the converted speech should be able to handle various language inputs. - The speech should be at a normal pace with a normal pitch, providing a natural and easy-to-understand listening experience. Ideal skills and experience for this job include: - Strong knowledge of the SMPP protocol and its implementation. - Proficiency in speech synthesis technologies and tools. - Experience in working with different language libraries and frameworks. - Ability to configure speech rate and pitch setting...

We request the customization of the linphone app for the Android, iOS, and Windows platforms, incorporating our logo and company information. Additionally, we require integration of the apps with our existing ASTPP PBX server, with the following desired features: Support for audio calls with various codecs (g729, ulaw, alaw, Opus, iLBC, GSM) Video calls with video preview, supporting codecs such as H.264, H.265, and VP8 Management of multiple calls Call transfer, pause, and resume functionality Audio conferencing by merging calls into a conference Instant Messaging with end-to-end encryption for both one-to-one and group chat Capability to share pictures and files Address Book integration Call History display Presentation of advanced call statistics Echo cancellation fe...

Looking for a sip soft phone that can connect to asterisk and make calls. It will use codec g729. Call history, settings and other standard features.

c++ work with githup lib to sample project iax2 using i need create sample project console application that can i send it iax2 username and password and server ip to check if register true . save info in file result And I want to add an option to run the program if I want to send a call With add timer to kill call and kill process from task manager

- Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to be demonstrated. Note: please read first & check android condition, if need root version use i thinking it. First you show me demo then add money freelancer website.

Looking for a sip softphone for windows desktop with g729 codec ,call transfer and conference option. Should work behind nat. Stun and rtp ports should be auto filled and static. We will build with basic options now and will upgrade more , so bid with your best budget.

Deseamos desarrollar una app android que haga las veces de softphone utilizando WebRTC. El softphone deberá tener una vista de log...que haga las veces de softphone utilizando WebRTC. El softphone deberá tener una vista de login, para lo cual usará un API REST de nuestro sistema de callcenter para validar sus credenciales y a través del cual recibirá los parametros para registrar su softphone WebRTC. Debe contar con funcionalidades de llamar, contestar, transferencia, conferencia, mute, hold, historial. Deberá soportar codecs g729, g711, opus. Además de la vista de softphone deberá tener una sección adicional con acceso a 03 reportes. Para obtener información de los reportes se brindará API REST el cual rec...

- Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to be demonstrated. Note: please read first & check android condition, if need root version use i thinking it. First you show me demo then add money freelancer website.

I need to compile the latest version of linphone android, you need to rebuild the SDK with the codecs g729.

Full VoIP on flutter (Fonctionnalities) - Audio Call - Video Call - Instant Messenger - Interconnected With Magnus Billing user account - File transert - Video Recording -Audio Recording Codecs - G729 - G723 - G711 - GSM Protocol - SIP - H323 - TLS - ZRTP Platform - Android - IOS - Web

I need my own softphone with standard codecs and codec g729, g723, video call, chat, call log, for desktop windows, mac, android and iphone.

VoIP/Sip client Android APP first of all the app must be fully encrypted. this app must me done with the SIP protocol standards. Basic Features: Incoming Calls Outgoing Calls Mute Calls Hold Calls Transfer Calls Integrated Call History Contac List change CallerID before calling or fixed optional: Send random caller ID before call. Advanced Features: voice/speech changer inte...CallerID before calling or fixed optional: Send random caller ID before call. Advanced Features: voice/speech changer integrated to the calling app Multiple SIP accounts setup SIP Account information hidden after initial setup DTMF (telephone tone) support using SIP INFO or RFC 4733 Provisioning via QR Code SIP TLS transportation SIP UDP transportation SIP TCP transportation Audio codecs: G711, G729,...

...calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via WIFI and an incoming and outgoing call would have to be demonstrated. ...

I want to enable G729 codec in my linphone desktop app project more information in chat

Hello I need an android developer to enable G729 codec in the linphone project android

I need an ios developer who can enable G729 in my linphone application.. here are some links that can be used.. i will share source code once the developer confirms he can do it

Hello The developer has already reskinned and did all requested..but left with G729 codec enabled pls expert on this issue should bid..i need it fixed within 24hrs..thanks

I need Need VoIP Softphone Source Code. I need support Android studio Latest Version and support voice code g729, g722 & others. If you have any ready source code then give me message.

i need Enable g729 on Linphone Android. i have use Linphone Latest Sdk

...viewing Full-Screen Video Conferencing Instant messaging and presence using the SIMPLE protocol Session timers Network diagnostics Auto Answer and Do Not Disturb Modes File transfer (compatibile with any SIP server) File sharing (compatibile with any SIP server) Remote Desktop over SIP Fax (beta version) Call and Chat History Audio and video recording Audio Codecs: G.711-Alaw, G.711-uLaw, G.723.1, G729, GSM, iLBC, L16, Speex, OPUS Video Codecs: MPEG1, MPEG4, Theora, DIV3, MJPG, H263, H264, VP8 WideBand and Ultra WideBand codec (speex, opus) Audio tuning wizard Dynamic Jitter Buffer Packet loss concealment (PLC) Automatic Gain Control (AGC) Acoustic Echo Cancellation (AEC) Voice activity detection (VAD) Noise supression Push notifications Auto QoS Dynamic Threshold Algorithm for ...

Hi, I need an IOS developer who has experience in working with Linphone, I want you to compile latest version of Linphone with G729 and customise it same as our existing android app. Here is the existing app: Please bid only if you can work with 100% dedication as we need to finish this very quickly. Thanks Vidhya Sagar Dixit

Se solicita desarrollo de softphone que soporte los siguientes protocolos sip alaw ulaw g726 g723 gsm g729 El server SIP, puerto SIP deben estar embebidos sin posibilidad de edicion por parte del usuario. El usuario solo deberia ingresar su usuario y clave SIP o permitirle escanear un codigo QR que le enviaria el prestador

We are a telecommunication company in Turkey which mainly operates in cloud pbx area. We ...we are going to need default registration method for international use) • QR Code – We need to implement QR code for easy registration and configuration steps • Config file or URL – Same logic as QR. We could use this on computers. • Video Call Support – (H264) • Admin Panel – We require this for all platforms. It will show usage statistics and we must able to change configuration of all users • G729 Support • Language Support – We will require multiple languages when we ready to enter the global market. • PUSH Notifications – This is one of most important requirements for us. Users must be able to take calls even if appli...

We are a telecommunication company in Turkey which mainly operates in cloud pbx area. We ...we are going to need default registration method for international use) • QR Code – We need to implement QR code for easy registration and configuration steps • Config file or URL – Same logic as QR. We could use this on computers. • Video Call Support – (H264) • Admin Panel – We require this for all platforms. It will show usage statistics and we must able to change configuration of all users • G729 Support • Language Support – We will require multiple languages when we ready to enter the global market. • PUSH Notifications – This is one of most important requirements for us. Users must be able to take calls even if appli...

We are a telecommunication company in Turkey which mainly operates in cloud pbx area. We...we are going to need default registration method for international use) • QR Code – We need to implement QR code for easy registration and configuration steps • Config file or URL – Same logic as QR. We could use this on computers. • Video Call Support – (H264) • Admin Panel – We require this for all platforms. It will show usage statistics and we must able to change configuration of all users • G729 Support • Language Support – We will require multiple languages when we ready to enter the global market. • PUSH Notifications – This is one of most important requirements for us. Users must be able to take calls even if applic...

Basic Linphone App with Customization with Own branding App to App calling App to mobile calling Multiple calls management Call transfer, pause and resu...conferencing (merge calls into a conference) Instant Messaging Pictures and files sharing Address Book Call History Display of advanced call statistics Echo Cancellation Quality of Service Secure communications: zRTP, DTLS, SRTP Bluetooth headset support Languages: English, French, Japanese, Arabic Account creation assistant Dedicated tablet user interface Audio  using codecs such as G722,G729,SILK, SPEEX Video calls using codecs such as VP8, H264, MPEG4 Integration with push notification (requires compatible SIP server, sip service has push enabled) Peer to Peer Audio and Video calling using ICE Sup...

I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: You will probably also need to check the G729 codec configuration

Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (Dinstar gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive,...

I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format F...GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all ...

in my asterisk moh files are played but cannot hear anything using g729 freelancer to help resolve this issue.M budget is 20$ for this ,please do not bid high i will not entertain such bids.

need expert to insatll g729 codec for pass calls.

We need an application that runs on Android phones (4.0 or above) and can receive calls through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please l...from SIP to GSM - Choose GSM module to use - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Server - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Be able to store a CSV with all ...

We need an application that runs on Android phones (4.0 or above) and can receive calls through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please l...from SIP to GSM - Choose GSM module to use - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Server - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Be able to store a CSV with all ...

i want to make a SIP dialer dialer have to work all network codec must be g729

iam looking for an android app which is similair to silver dialer/i tel dialer (referrence ) which is compatible with g729 audio codec & works with all standard soft switches. works well with minimal bandwidth

We are looking for solution like a traditional GSM or CDMA VoiP gat...and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance th...

i want to convert my android phone as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. (VOIP CALL TERMINATION) so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. an...android phone as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. (VOIP CALL TERMINATION) so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on ...

I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: I may probably also need help with G729 codec configuration

I want to build an optimized image of the latest version of Asterisk for the latest Alpine Linux stable. The final product is the Asterisk compiled and tested with load in an AWS server and Docker. The system should have the codecs g729, opus and SILK compiled. The AWS image should be able to run without docker The Sources should be available to recompile minor updates

Need to build...Enable to route calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to be demo...

I need an...calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a seperate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to to be dem...

Hi, We are using asterisk server and g729 codec. It supports only Pass-Threw mode because it's the free version. But we need a free version of this codec that supports Transcoding mode. If anyone have that please knock me. My budget is below $50 Thank you.

Hi, We are using asterisk server and g729 codec. It supports only Pass-Threw mode because it's the free version. But we need a free version of this codec that supports Transcoding mode. If anyone have that please knock me. Thank you.

Hello, IF YOU DO NOT KNOW WHAT LINPHONE IS THEN DONT READ FURTHER. We need experienced linphone iOS developer who has readymade solution to compile linphone iOS with G729 codec. Linphone version 4.3 You will have to download open source code from linphone website and compile with G729 code and pass me the compiled source code with detailed steps. Please only apply if you have done it before. Thanks

Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) A. Any mini Linux distribution exam- puppy Linux , linux mint B. Fedora desktop distribution C. Centos 5.8 or 6 7. Server A to Server B voice traffic will be e...

Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and t...

We need an application that runs on Android phones (4.0 or above) and can receive calls through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please l...from SIP to GSM - Choose GSM module to use - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Server - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Be able to store a CSV with all ...

I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK form...GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all ...