Softphones srtp jobs

...C#, or any other compatible with the Milestone SDK. Tools: Milestone SDK, SIP libraries such as PJSIP or Linphone. Compatibility: The SIP client must be compatible with current Milestone versions. Support for standard SIP protocols and common audio codecs such as G.711, G.729, etc. Security: Implement security measures to protect the SIP client configuration and data transmission. Support for SRTP (Secure Real-time Transport Protocol) encryption and TLS (Transport Layer Security) for SIP connections. Work Plan Design Phase: Initial meeting with the development team to define detailed requirements. Design the architecture of the SIP client and its integration as a plugin in Milestone. Development Phase: Develop the SIP client and configuration interface as a plugin. Implement ...

Project Description: FreePBX and GOIP Configuration Looking for the initial setup of a FreePBX systems with GOIP GSM gateway for inbound trunks. Softphones will be used by internal clients to receive calls. Skills and Experience Required: - Experience with GOIP GSM gateways. - Proficiency in FreePBX configuration. - Softphone configuration. - Understanding of custom configuration setups, if necessary If you have the expertise in these areas, please bid on this project. Thank you.

I am looking for a freelancer who can help me set up and configure my Twilio account. Here are the details of the project: Setup twilio account for using soft phones Simple IVR menue Sip trunk Preferred Communication Channel: - Both SMS and Voice Call Workflow: - I have a detailed plan that I want...to implement with Twilio Technical Level: - I have an intermediate level of understanding with Twilio Skills and Experience Required: - Experience in setting up and configuring Twilio accounts - Knowledge of Twilio's SMS and Voice Call features - Ability to understand and implement a detailed plan - Strong communication skills to collaborate and finalize the workflow - experience setting up softphones/ sip routing If you have the necessary skills and experience, please apply fo...

...Server Skills and experience required: - Strong knowledge and experience in hosting VoIP softphones on Windows servers - Proficiency in configuring and managing VoIP protocols, particularly SIP - Ability to handle concurrent calls ranging from 1 to 10 Project details: We are looking for a skilled freelancer who can help us host a VoIP softphone on a Windows server. The server will be running on the Windows operating system. Key requirements: 1. Operating System: Windows 2. VOIP Protocol: We are open to suggestions as the client did not specify a preference. 3. Concurrent Calls: The softphone should be able to handle 1 to 10 concurrent calls. If you have a strong background in hosting VoIP softphones on Windows servers, and are proficient in configuring and managing VoI...

...functionality i need for a virtual receptionist which vonage business has (Ring to user before going to Virtual Receptionist & Call Tagging). Vonage does not have the outbound capabilities I need (Automatic Local Presence for Calls & SMS & a Power Dialer). So I'm trying to figure out a way to use both inside Salesforce, but am not sure what is possible. 1st idea is creating 2 separate open cti softphones within salesforce. One for all outbound calls (click to dial, dial pad, and power dialer) as well as all SMS features (360 CTI/SMS will have this part covered). The 2nd softphone (Vonage Business Center or possibly Vonage Contact Center) would handle all inbound calls so that I could have a ring first to & call tagging feature for the inbound call while ...

...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 feature Quality of Service optimization Ability to send and receive SMS text messages through the ASTPP PBX server Implementation of secure communications through zRTP, TLS, and SRTP Support for Bluetooth headset devices Language options: English, Chinese Integration with ASTPP PBX server to check balance Integration with ASTPP user accounts Message Archive Management (MAM) functionality Implementation of PUSH Notification from the ASTPP PBX server (FreeSwitch) Customization of the app with our logo and company information Provision of source codes. K...

VoIP Recording - SIPREC Re-Invite and Audio...VoIP system and I need assistance with SIPREC Re-Invite and audio streaming. - I require live streaming of the audio for real-time monitoring and analysis purposes. - There is no need for transcription of the VoIP audio stream. Skills and Experience Required: - Experience with SIP VoIP systems and SIPREC Re-Invite is essential. - Proficiency in audio streaming technologies and protocols. RTP and SRTP - Knowledge of real-time recording and live streaming techniques. - Familiarity with VoIP monitoring and analysis tools. - Attention to detail to ensure accurate audio streaming without transcription errors. - The code implemented could be in .Net Core, Python or Java. - Knowledge of Acme or Sonus SBCs, and Avaya, Cisco and Genesys is a di...

I need a webrtc to sip gateway to be implemented so I can connected some webrtc softphones (asterisk webrtc softphones on our Odoo CRM) to twilio sip domain

- Install fusionpbx - Setup Inbound/outbound SIP trunks - Setup 3 extns - Test softphones - Delivery

I need an Android app similar to Zoiper, for business use. The app should have basic call functionality, including making and receiving voice and video calls. Additionally, the app should have a modern design that is easy to use. The ideal freelancer should have experience in developing VOIP SIP softphones for Android.

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

...protocol and SIP states. Your task will be to help us to identify things on SIP to be able to discuss with our developers (with low expertise on SIP). The developers have to implement some features, but do not understand the SIP protocoll well to find the correct paths. Your task will be to help in the discussions about low level SIP featuers like: - how to list all registered SIP devices (softphones, smartphones-apps, desktop-apps, physical phones, ...) - how to identify how many onging calls are running in parallel? - how long is each call onging? - who (device) has taken the call, what time ended the call, ... - and many more Your task will be to consult only, except you are a developer too. If this is the case, you are welcome to join too after going through a interview ca...

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

...instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL like The full SIP environment will be: - your SIP app, connected to the sipgate-account - optionally other machines also running your SIP app on multiple desktops - one or more SIP physical desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account Process: 1. your SIP application shall detect incoming POTS calls (rings, no hook take off) 2. if possible to detect which device has taken off hook (the call) (e.g. if the desktop phone picked the call) 3. if possible how long the call was going with the different phone to measure the call duration The listener shall be configurable via

We are looking for an experienced VoIP developer who can design Windows and MAC desktop VoIP applications using our Hosted PBX API. The application will have to be tightly integrated with our asterisk-based PBX and our custom API. Supported functionality will include: Voice calling via SRTP Searchable Call history with access to call recordings and call notes SMS and MMS messaging Read-only access to favorites and BLF keys Read/Write access to personal contacts Visual Voicemail Do not Disturb Call Forwarding We prefer a web application running installable with an Electron wrapper on the client's workstations but are willing to entertain other options.

...fully working sip account after award) The caller-URL contains some query parameters, like the callers number. The app can run only one instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL. use jitsi desktop event handling The environment: - your SIP jitsi app - one or more SIP desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account 1. your SIP application shall now detect incoming calls (rings) 2. if possible to detect which device has taken off hook (the call) 3. if possible how long the call was going with the different phone The listener shall be configurable, means - what URL to call - which phone numbers to ignore (regex) - which phone numbers...

...caller-URL contains some query parameters, like the callers number. The app can run only one instance and is listening to incoming calls. On incoming call it triggers a browser open of the above caller URL. use or similar java library/framework The environment: - your SIP application to implement - one or more SIP desk phones running on same SIP account - one ore more softphones (laptop, smartphone, ...) running on same SIP account 1. your SIP application shall now detect incoming calls (rings) 2. if possible to detect which device has taken off hook (the call) 3. if possible how long the call was going with the different phone We have some exiting code of a proof of concept, which does not work anymore. So you do not start from scratch. But it is mandatory that

...SIP provider only. The caller-URL contains some query parameters, like the callers number. As a UI-less (cmd line only) app on the desktop it will be so a simply app which does basically a call listening and triggering a browser based URL to show. The environment: - one SIP account which has a real phone number - your SIP application to implement - one or more SIP desk phones - one ore more softphones (laptop, smartphone, ...) 1. your SIP application shall now detect incoming calls (rings) 2. if possible to detect which device has taken off hook (the call) 3. if possible how long the call was going with the different phone We have some exiting code of a proof of concept, which does not work anymore. So you do not start from scratch. But it is mandatory that you are a master...

Asterisk full setup in our enviroment ( 10 user , 6 cisco phones). Full setup as a virtual pc on Synology NAS. Setup for softphones on mobile devicese and connect existing cisco hw spa504g phones + call recording etc ...

we are looking for a very experienced ...are a bad fit, if you have never done tasks of integration SIP into Java A simple test task for seniors in SIP+Java: - listen to incoming calls (SIP provider can be provided to you) - trace incoming calls to sysout with the phone number Milestones: MS1: very basic implementation of SIP MS2: the rest of the above named requirements MS3: with and without TLS / SRTP support TLS / SRTP, shall be mandatory for later usage of this app budget? will not be disclosed, place your best bid timeframe? starting in the next 2 weeks time commitment? about 1-4h a week for the initial consultancy. If you want to get incorporated into development, you will be awarded tasks, which you estimate, and we award you afterwards the implementation

...JUnit (no UI) What is NOT needed: - a UI (not required, implement a JUnit test to call your functions) - a service architecture (like spring or JEE) - any persistence - a voice/audio implementation for SIP (only incoming ringing required) Milestones: MS1: very basic implementation of SIP MS2: the rest of the above named requirements MS3: with and without TLS / SRTP support TLS / SRTP, shall be mandatory for later usage of this app What are our requirements? - your code passes checkstyle, pmd and spotbugs (we will share you a git repo with eclipse settings) - JDK17 - maven - 24/8 formula - create a model class representing the input of your function - create a service class implementing the logic - create a unit test, which tests the service class - we do NOT need

I need someone to review my TLS & SRTP configuration of the Kazoo VoIP cluster software. I am having issues communicating between my Kazoo cluster install and my SIP trunk provider. The error message between returned on outbound calls is 488. The inbound calls don't appear in logs of my Kazoo cluster.

I need someone to review my TLS & SRTP configuration of the Kazoo VoIP cluster software. I am having issues communicating between my Kazoo cluster install and my SIP trunk provider. The error message between returned on outbound calls is 488. The inbound calls don't appear in logs of my Kazoo cluster.

Dear Amrit, we changed the login configuration of the mobile application and this project is catered to support the work of this configuration. With the configuration, we will look at the app enabling SIP TLS and SRTP. We also added the domain setting so that user can change that domain setting. Thank you. John

i need help to setup my gsm gateway into one login id and password for softphone so that we can receive calls on the move

...including GUI. PBX shall be deployed on premise on server/hardware. It will be limited to number of users/phones and license to be uploaded/entered for upgradation of users/phones. GUI will have company branding. PBX software shall be inclusive of below modules but not limited to these only. - Detailed CDR - Call Recording - IVRS and log - Phone Logo change - Live Monitoring and real time status - Softphones integration - Video Conferencing - FXO/FXS/PRI/Trunk Gateway Integration & support - License based features to limit number of users/phones - Free Modules of freepbx/Pbxact - Built-in Redundancy Developer need to provide 6 Months support in this project from developer for testing/bugs/reports/problem of any kind. Future upgradation shall be possible on the software and s...

...aggressive goals and high-quality standards. Creative approach to problem-solving. Ability to develop long-range project plans and schedules. Experience in embedded software and firmware design and development. Bonus Points: Comfort with tools such as debuggers, logic analyzers, and oscilloscopes. experience in Embedded C/C++. Experience with network programming and protocols (TCP, UDP, HTTP, SRTP, etc.). Experience in working up and working with hardware-focused communication interfaces (MIPI, SERDES, I2C, RS232, USB, Ethernet) Experience in working with HAL Firmware development experience on MO, M3, or M4 embedded cortex Development in embedded C/C++ in IAR, Mbed, or Keil Development Environments Thorough knowledge of Embedded System Architecture with RTOS. Prior experience ...

We need a Technical person who can install a functional VICI Dialer on a server of his choice and knows how to configure softphones like Zoipher, eyebeam to the Dialer for US Calling. He will be paid for Installation as well as maintenance of the dialer very month. He must know about AC-CID and the Outgoing call must be that of the Customer. Must have expert knowledge of Installing VICI Dialer and Configaration

Install TLS 'let's encrypt' certificate, setup PBX to use TLS with SRTP

Dear all, I am seeking mobile APP and software developer to integrate ZeroTier into LinPhone softphones (Mobile APPs ANDROID & iOS & desktop applications for Windows and MAC) . both are opensource and it should be straight forward task , thanks for the available documentations for both , you may find details on The integration part: Is simply making ZeroTier embedded into LinPhone ZeroTier should work inthe backgroud upon launching, and would ask for the "Network ID" Upon adding the "Network ID" , by default it the connection estabish shall be pending till I authorize the connection "ZeroTier Address or AKA Node ID" from my end, (this connection

I am using Elastix 2.4 call center module. I'm not having any problems. I just can't install WebRCT. That's why I'm looking for an alternative solution. Things I need; A stable working call center module. WebAPI support for call center (I'm using Elastix ECCP) I want API support like this. I want WebRTC. I don't want my employees to use softphones like Xlite, Zoiper.

I want a professional looking flyer to market my telecommunications business. I want it to have 5 sections with photos and description 1. Data cabling 2. Internet Services and Onsite PBX 4. Headsets Voip Desk Telephones and Softphones 5 Camera Systems for Businesses

I want a professional looking flyer to market my telecommunications business. I want it to have 5 sections with photos and description 1. Data cabling 2. Internet Services and Onsite PBX 4. Headsets Voip Desk Telephones and Softphones 5 Camera Systems for Businesses

We need a WebRTC client SDK that can be implemented in 3rd party projects. Needed functionalities: WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/d...chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/disable video during a session (during an ongoing call session re-INVITE and disable or enable video) WebRTC signaling plane: - SIP over WebSecureSocket (will connect to a sip server as Kamailio/Opensips/FreeSWITCH) WebRTC media plane - codecs: 711, opus, VP8, VP9, H.264 - DTLS/ICE/SRTP API facing side: - provide an easy and comprehensive API for quick integration into 3rd party...

Set up Asterisk PABX and integrate with proprietary CRM. Asterisk to have web interface to manage DDi routing and ring groups. API for CRM integration will be available. An option to switch easily between desk phones and softphones

I need a Sip Client Webphone so I can integrate this into my CRM as click to call button for my clients so basically my agents will see the clients details and when they click on the call button a call will initiate to the client number via Sip Client Webphone and WebRTC and right in the CRM agent talks with the client. I don't want normal integration to other platforms and softphones like zoiper, 3cx or others I want something like: but for some reason these are not working for my sip they asking AOR or WSS which my PBX/VoIP doesn't provide so it should connect to sip only using username, password, and domain of PBX like zoiper connect.

Hello, we need a platform where we can launch our VoIP business. **** This is second attempt to get this done.. I will give you the full details of my last experience. I will give you access to what we already. If you can fix i...second attempt to get this done.. I will give you the full details of my last experience. I will give you access to what we already. If you can fix it,, then great.. If not, then we need to either build our own system.. Or use a White label Cloud PBX solution.. -We need complete billing platform. ( I don't want to walk you through what is needed in a working functional VoIP Billing platform. -Softphones tied into the billing platform.. -Easy customer onboarding Please ping me, only if you have experience in doing such a job.. And you have sample wo...

We are currently using a custom PBX solution. The PBX server is hosted in the Vultr Cloud and is based on Freeswitch with the frontend as FusionPBX. There are few issues which have come up which needs to be fixed urgently. The SIP trunk is being provided by Voxbeam 1) All incoming calls should ring the SIP softphones 2) Outgoing is not working properly for some regions. Need to diagnose that and resolve 3) If possible, need to configure an alternative SIP trunk in case Voxbeam is failing. The process should be automatic.

This project is the installation and configuration of fusion PBX on Debian 10. You will be provided access to a VM. What must be delivered Full functional system integrated to 1 sip trunk, VoIP phone and softphones A fully secured system that is only accessible to admin and users registered in PBX Full documentation of how to in Freepbx eg: add new PBX user, configure softphone, setting to be used for softphone, VoIP phone config and settings Auto-attendant with MOH for inbound Leave a VM by pressing a number VM must be sent as an email to admin and PBX registered users. New users can only be added by the admin Outbound calls are restricted to users registered in the PBX. Call recording when a call is answered. You must have experience of doing so as well. The development must ...

...for each call received (see Fig. 1). MapMaker is based on a FileMaker Pro (version 19.2.2) database. A balloon for a marker on the map (Fig. 1) contains a phone number which can be clicked to ring the person or organistion. This is achieved by simply clicking on the phone number. When clicked, a third-party softphone is used to ring the required contact (see Fig. 2). Unfortunately, proprietary softphones contain many more "bells and whistles" than required, evidently cannot provide the desired feedback to MapMaker and are proving to be excessively costly. It is therefore required that MapMaker be modified to avoid the need for a third-party softphone. PROJECT SPECIFICATIONS When a phone number is clicked in MapMaker, it is required that the following parameters be mad...

Configure Twilio For SIP and Configure 2 Softphones and 1 SIP Algo 8201 Device

Dear All, I have a fresh installed FusionPBX on a Cloud [Ubuntu 20.04] and I have configured it to use TLS And the remote registered extensions haven't audio Note: the remote registered extensions are behind a NAT and are located in a country is restricted the SIP traffic so I should use TLS with SRTP to bypass the SIP blockage. Note: Award will be to the one have an experience only to complete this project successful (no wasting time) Thanks to all who will bid on this project!

Hello! I have installed a Cloud FreePBX server [Fully configured] SIP with TLS + SRTP But the user's location is in a country that has SIP Traffic restrictions (Blockage) Everything is working and the calls go through 2 SIP Extensions but there is something wrong with the voice! The voice is like shaking! Note 1: If I use a VPN to avoid the SIP traffic restrictions, the voice is so clear and there is no such problem in the call Note 2: The funny is once I try to dial the other extension of *43 for the Echo test and our voice is still shaking If I pressed any num key twice or three times, our voice becomes so clear! I think may one of the following ideas are resolving the problem: 1- Install & Configure a Cloud SIP Proxy Server like Kamailio, OpenSIPS or any other SIP Pr...

...understanding of network protocols and associated setup required. Skills required: • Hands on experience in JavaScript, Java, C++ etc. • Extensive technical knowledge and working experience in Data transport services, technologies. • Work experience and knowledge in SIP based telephony solution (VoIP). • Working experience on Datacenter, Cloud environment. • Deep knowledge of protocols such as SIP, RTP/SRTP including TLS, webRTC • Very good knowledge of protocol and packet analysis • Experience working with switching and routing protocols • Strong understanding of NAT traversal for VOIP (incl. STUN/TURN) • Experience working with Linux and Windows server operating systems • Knowledge of third party/open source SIP stacks • S...

Hello, I work as a volunteer in an emotional support and suicide prevention NGO. We get about 10,000 calls every day, in which people let off steam for about 30 to 90 minutes. We currently use softphones installed on the computer but I would like to place a SIP proxy and run a webphone in the browser. For that I will need an executable docker-compose file or detailed instructions for: - Frontend - inputs SIP parameters as ip, login and password - button to connect - disconnect button - button to answer the call - button to end the connection - no need to make calls, just receive calls - it doesn't have to look nice, I will adapt the frontend to our internal application. - you can use sipjs, jssip or any other open source SIP library. - backend a backend ...

Hi, I need support setting up a basic on premise (RasPBX) Asteriks installation. I have the following devices: - Raspberry PI 4 with RasPBX installed - 2N technolgies Doorbell with Sip Support (works already and is registered to s...following devices: - Raspberry PI 4 with RasPBX installed - 2N technolgies Doorbell with Sip Support (works already and is registered to sipgate) - Several Devices with Sip Softphone installed - Home Assistant on several devices (I want to connect via webRTC (Browser) to the Asteriks) So I need: - basic configuration of Asteriks / FreePBX - 2-3 extensions / endpoints for the devices / softphones - 1 extension for the webRTC connection (e.g. ) - 1 configuration for the 2N Doorbell with SIP Support

To set up installed Asterisk PBX stable 16 qnap x86 SSH small set up 4 sip 5 phones 5 softphones GUI - web admin simple voice missed number to email debatable option

I require somebody to take a look at a new VitalPBX install and fix a few small issues I'm experiencing. I've configured trunks, extensions, dial plans etc however I'd like the following things configuring/setting up: - Extensions via TLS - configure device profile for TLS - VitXi - possibility to use push - SRTP working

Hello all, I have Freepbx installed on a VM with a public IP address. I am testing with 2 softphones which can register with PBX and call each other. But there is no audio at all. I would like your assistance to debug and fix the audio issue. I will happy to discuss my setup with shortlisted candidates.