WebRTC to SIP gateway

$30-250 USD

Em Andamento

Publicado

há mais de 4 anos

$30-250 USD

Pago na entrega

I need to set a gateway that will be use as a proxy between Asterisk server and web clients. User will log to the gateway and the gateway will connect it to the specified server with SIP user and password. I'm expecting to get the server installation process and code with the client side code that provide credentials login. Once client will connect he'll be able to call and get calls using his browser. The gateway will have SQL db that contain the user credentials to connect and the SIP credentials to register to the asterisk server

ID do Projeto: 21728499

Sobre o projeto

9 propostas

Projeto remoto

Ativo há 4 anos

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9 freelancers estão ofertando em média $200 USD for esse trabalho

@Erewin

Hello, i can make it for 350. I have done this based on Kamailio. be free to ask me any question. bye, Pavel

$350 USD em 14 dias

5,0

(69 avaliações)

6,7

@mula9211

Hi, I have 15 years of experience in VOIP and asterisk. I can demonstrate my working model for the same scenario. I offer my services for your project. Please have a look at my profile to see all my experience details. I assure you of a high quality professional service. We can chat out details.

$200 USD em 7 dias

5,0

(52 avaliações)

6,1

@volga629

NetworkLab is experts in VoIP we can help you deploy proxy layer for webrtc. Our reach experience will help complete the project quickly and in professional manner.

$240 USD em 4 dias

4,9

(53 avaliações)

6,2

@thinkbased

Hi , the new versions of Asterisk is able to connect to web socket in order to have calls . I can setup and make it running if you are interested.

$200 USD em 3 dias

5,0

(11 avaliações)

4,5

@GlobalSolution3

Hello I am full stack developer with 10+ years of experience in web/mobile development. I am familiar with WebRTC/WebSocket. Lets discuss more details via chat. Thank you.

$140 USD em 7 dias

5,0

(12 avaliações)

4,0

@saadjehangiri

I am currently working on different voip projects both at client end and sip provider end. I have more than 6 years of experience in VOIP solutions and more than 8 years of experience in Linux and Networking. I got asterisk professional training and certification from authorized Digium Center in Dubai. Asterisk/Freepbx SIP/IAX/PRI trunks Asterisk /Freepbx twilio elastic trunk Asterisk /Freepbx call termination and origination Asterisk/Freepbx/Elastix/ dynamic ivr using text to speech engine and save DTMF feedback of users in Database.. Asterisk/Freepbx Dialplans Asterisk/Freepbx IVR , Automatic Call distribution Asterisk/Freepbx Queues Asterisk/Freepbx dynamic logins Asterisk/Freepbx Call Recordings Asterisk/Freepbx Migration to different cloud technologies. Customization of asterisk based Voicemail service Asterisk/Freepbx based cost effective voicemail development using IBM Watson Speech Recognition API Realtime asterisk deployment CRM integration with asterisk Speech recognition API Text to Speech API Elastic Server configuration trunks, DID, inbound and outbound routing Elastix server troublshooting and support Wazo installation and configuration Wazo server deployment Wazo and IBM watson api integration Wazo endpoint configuration Wazo Trunk deployment Wazo and Sip trunk Carriers deployment Freeswitch Installation and Configuration Freeswitch/Fusionpbx multitenant deployment for Telecom operators Freeswitch/Fusion Sip profiles Freeswitch/Fusion dialplan

$150 USD em 7 dias

0,0

(0 avaliações)

0,0

@quansoft

Dear Sir, I am offering, Asterisk/PBX/Freeswith to develop VOIP dialer, SMS broadcast, ringless voice message, custom CMS with CRM using codeigniter/bootstrap. I will share the system once you will connect with him . I look forward to hearing your response soon Regards pankaj

$140 USD em 7 dias