Getafreelancer srtpεργασίες

...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 ...

...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...

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.

...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 calling of browser open with: windows: start "" "" linux and mac too 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

...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

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

...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 ...

Hi there, I am looking for an experienced .Net software developer to create integration tests for an existing net6.0 ASPNetCore project. I want the usage of WebApplicationFactory to follow these instructions: The project has around 6 ...com/en-us/aspnet/core/test/integration-tests?view=aspnetcore-6.0 The project has around 6 controllers with around 50 GET/POST endpoints. I want you to create basic happy path tests for all 50 endpoints. Around 10 services will need to be Mocked and injected into the WebApplicationFactory. The solution will be in a private git repo which I will give you access to. This is not the usual spam - i'm an old GetAFreelancer account looking for simple, serious contractors.

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

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...

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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...

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

Basic Linphone App with Customization with Own branding App to App calling App to mobile calling Multiple calls management Call transfer, pause and resume Audio 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 ...

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Preciso q o sistema suportar a criptografia dos tráfegos de sinalização e de voz por meio dos protocolos TLS e SRTP para conexão dos ramais aos softphones e ou telefones ips

I have a project where I need a native german or austrian to speak the text I am providing. I need a quote per spoken 2-3 A4 page to Audio (unkompressed such as WAVE or Mp3)... translate german text reading, words german text, german text outsource, text to speech daniel, text to speech voices, kor...(unkompressed such as WAVE or Mp3)... translate german text reading, words german text, german text outsource, text to speech daniel, text to speech voices, korean text to speech, dj name voice maker online, text to speech mp3 with natural voices, text to speech google, text to speech robot, text to speech free download, english, audio services, voice talent, german, audio production, voice over, getafreelancer german text, german text english translation freelance, freelancer german ...

Hi Mohammad Saydul K., I noticed your profile and would like to offer you my project. We can discuss any details over chat. The project is about to integrate Teams, Kamailio and a SIP provider and asterisk. Teams and SIP provider registration is already working, including certificatates etc. The major part of the work is to configure Kamailio to rewrite phone numbers and make RTP, sRTP work.

Hi There, This logo is for a Gourment Company, "SRTP Gourment". Company Name is "SRTP GOURMENT" FULL NAME SHOULD BE THERE IN LOGO This company sells spices, Seafood, and many other raw eating materials to be used in the Kitchen. Need a logo to be perfectly seen on a dark and light surfaces.

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Hi Sergey S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. What I want you to do? Create PBX server with SRTP, ZRTP and voice pitcher: I will rent OVH server, so thats not a problem. But I want to have fully encrypted VoIP Calls. Tell me price, and time.

...Javascript, Typescript, Node.js, WebSocket, OpenSSL, Protobuf, pouchdb. 4. DevOps, system integration and support specialist: Total experience 12+ years, experience with o Linux operating system (Redhat, CentOS) o Network architectures and security (Firewalling and IDS) o System architecture for high availability and scalability o Networking and communication protocols (TCP, UDP, TLS, SSH, DTLS, sRTP) o Job automation using Ansible o Shell scripting (perl, python or/and bash) o OpenSSL command line and SSL/TLS certificates management o F5 BIG-IP and load balancing techniques (LTM & GTM) o Deployment and management of systems on cloud platforms like MS Azure, GCS and AWS o IT Service Management Tools (e.g., Jira) o DevOps and in particular experience with Jenkins, GitLab, Git...

...Javascript, Typescript, Node.js, WebSocket, OpenSSL, Protobuf, pouchdb. 4. DevOps, system integration and support specialist: Total experience 12+ years, experience with o Linux operating system (Redhat, CentOS) o Network architectures and security (Firewalling and IDS) o System architecture for high availability and scalability o Networking and communication protocols (TCP, UDP, TLS, SSH, DTLS, sRTP) o Job automation using Ansible o Shell scripting (perl, python or/and bash) o OpenSSL command line and SSL/TLS certificates management o F5 BIG-IP and load balancing techniques (LTM & GTM) o Deployment and management of systems on cloud platforms like MS Azure, GCS and AWS o IT Service Management Tools (e.g., Jira) o DevOps and in particular experience with Jenkins, GitLab, Git...

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We have Two Avaya SBC Servers with Two SIP lines and 2 IPO Office Ver11 Server edition. We need configure 2 SIP trunks on SBC and Its should work with IPO. Also, We need configure Avaya XI Workspace with TLS and SRTP with Public Access.

...firewalls (different networks than the server, with NAT), and one on the FreeBSD server itself. 2. A publicly available extension, like me@, that anyone can use to make a SIP call that connects through the server to a softphone, and lets the caller leave a voice mail if no answer (which would be emailed to me). 3. Everything secure: No unauthorized access. 4. Be able to switch between RTP and SRTP by changing the configuration. 5. If possible: no asterisk database connection, only flat files. If not possible, explain why. 6. Asterisk comes up and runs as cleanly as possible: no unexplained error/warning messages. 7. Use the defaults that come with asterisk installation where possible. No modifications that will hinder a "pkg upgrade". I cannot grant access to the serve...

... 10. Sms/mms sending receiving 11. Overview of messages by contact / phone number / sip ID 12. Individual message thread 13. Voicemail - listen, record greeting, delete ( downloadable to app when needed from signalwire and deleted in sync) See more : send sms, mms, receive sms,mms midlet, VoIP trial test calling, VoIP softphone skins for both iPhone and android phones, visual voicemail, srtp, use signal wire test account to build a complete solution for one number and ability to add more numbers in a same account or creat more signalwire account seamlessly to work with this app. Open h264 sip video. And all audio codecs for 3G,4g, Wi-Fi data. Call kit integration is a must ( even if a phone is sleeping - app must run in background to revive phone calls and text messages c...

...recent 10. Sms/mms sending receiving 11. Overview of messages by contact / phone number / sip ID 12. Individual message thread 13. Voicemail - listen, record greeting, delete ( downloadable to app when needed from signalwire and deleted in sync) See more : send sms, mms, receive sms,mms midlet, VoIP trial test calling, VoIP softphone skins for both iPhone and android phones, visual voicemail, srtp, use signal wire test account to build a complete solution for one number and ability to add more numbers in a same account or creat more signalwire account seamlessly to work with this app. Open h264 sip video. And all audio codecs for 3G,4g, Wi-Fi data. Call kit integration is a must ( even if a phone is sleeping - app must run in a background to revive phone calls and text mess...

User-friendly voice and messaging app with enterprise security. ZRTP vs SRTP No backdoors, no man-in-the-middle attacks Unlimited peer-to-peer calling with out-of-network calling Unlimited peer-to-peer calling Call Mobile or Landlines = VOIP Calling Secure messaging and file transfers Scheduled burn functionality on both ends Available for any Android or iOS device

We have the i...unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a public IP (direct). Endpoints are Acrobits sofpthone users (android/iOS) connecting via WAN. there is nothing in between. All end-users use TLS+SRTP. and Acrobits Push. So ping is huge sometimes. I suppose qualify option may be the cause here. Official FreePBX forum treads ignore the issue and ask to order their paid support. As the issue is in the production system, there is no place for experiments and we can work only via remote session. I have all the accesses required. The issue is urgent. Please ping ...

Build React Native VoIP App/SDK with SIP protocol which able to integrate with current React Native solution. Feature/Function that require as below: • Push Notifications • Receive VoIP calls on the lock screen, active VoIP calls are no longer disrupted by incoming cellular calls and VoIP calls are now stored in the native applications call log. • Call Transfer • Call Recording • SRTP Audio • Able connect to local PBX server • API parameters to be ready for made call in existing React Native Apps • Receive call duration upon end call

We have our own existing customers on pbx servers and for them I would like to make a mobile ap for ios and android. to make calls with all pbx functionalities These functionality we need for our mobile app for VOIP. • Push Notifications • receive VoIP calls on the lock screen, active VoIP calls are no longer disrupted by incoming cellular calls and VoIP calls are now sto...make a mobile ap for ios and android. to make calls with all pbx functionalities These functionality we need for our mobile app for VOIP. • Push Notifications • receive VoIP calls on the lock screen, active VoIP calls are no longer disrupted by incoming cellular calls and VoIP calls are now stored in the native applications call log. • Multiple Accounts • Call Transfer • Ca...

We're looking for a customized version of CSipSimple with xmpp and sip tls/srtp features and full remote provisioning system. the app should bind with server once with user/pass and form then receive all configuration by api and check frequently based on a time interval. VoIP application (android) - Based on opensource CSipSimple - SIP-TLS / SRTP encryption support - Remote configuration/provisioning o All sip settings, user settings, codec, stun and server settings should be obtained from api o based on user/pass o time based subscription limit o time interval to renew the setting frequently - Always running in background with sip registered. - Hide all setting and server/account information o Can't login or logout of account o Can't edit any account info - ...

I need a freelancer pjsip expert who can build PJSIP stack with SRTP support and TLS support.

We would like to have the customized linphone app with ...resume 5. Audio conferencing (merge calls into a conference) 6. Instant Messaging - one to one and group chat (End-to-end encryption for messaging) 7. Pictures and files sharing 8. Address Book 9. Call History 10. Display of advanced call statistics 11. Echo Cancellation 12. Quality of Service 13. Send and receive of SMS text message through ASTPP PBX server 14. Secure communications: zRTP, TLS, SRTP 15. Bluetooth headset support 16. Languages: English, Chinese 17. Intregate with ASTPP PBX server for checking balance 18. Intregate with ASTPP user account 19. Message Archive Management (MAM) 20. Implement PUSH Notification from the ASTPP PBX server (FreeSwitch) 21. customized logo and company information 22. provide so...

Hello. I need someone who would install IP based PBX system and A2Billing for its provisioning. Also transport channel needs to be TLS and SRTP.

We want someone to configure a Kamailio server on AWS(amazon services) to work with IPv6 and IPv4 both. we expect the following to work Stage 1 (with outbound proxy enabled ) 1.) IPv6 clients should be able to us...IPv4 sip servers 3.) IPv4 clients should be able to use the Kamailio server as a proxy and register to IPv4 sip servers 4.) IPv4 clients should be able to use the Kamailio server as a proxy and register to IPv6 sip servers 5.) TLS must be used 6.) clients from ipv6 only networks must register to ipv4 sip servers using this as the proxy. stage 2 1.) clients send the call with media encryptions SRTP ZRTP and DTLS, calls must go to sip server with the encryption. 2.) we need both IPv4 and IPv6 clients to use this. ALL the implementations must be done on the servers w...

SIP Client Application for Both IOS and Android prefer to go with Linphone Features : - Audio & HD video calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Instant Messaging with message delivery status (IMDN) Pictures and files s... Audio & HD video calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Instant Messaging with message delivery status (IMDN) Pictures and files sharing Contact list Call History Display of advanced call statistics Echo Cancellation Call quality indicator Secure user authentication : md5 / SHA256 digest, TLS client certificates SRTP, zRTP and SRTP-DTLS voice and video encryption Supported languages: English, ...

...TCP/IPv4/IPv6 DHCP Client Extensible Command Line Interface Web Server with Extensible Configuration Web App and Configuration and Control Web Services VoIP Call Manager (includes support for incoming calls, outgoing calls, hold, conferencing, transfers, mic/speaker volume control and mute, etc.) Configuration Subsystem (file based or can integrate with other platform standard) Optional Security (SIPS, SRTP) Example Applications include: IP Speaker Phone IP Intercom Systems IP Ceiling Speaker IP Elevator Communications/Music IP Paging IP Door Entry IP Emergency Phones IP Parking Garage Entry IP Audio Streaming I need to give a complete solution, including software and hardware. cpu must use the mainstream arm or dsp chip（）, easy to buy, the price is cheap (less than 7 dollars)....

HI, I need to make linksys pap2 work via SRTP on my voipswitch. The ATA (client / pap2) should register to my switch via a proxy server i.e. SRTP server. I would need setup and training on how to do so in future.

HI, I need to make linksys pap2 work via SRTP on my voipswitch. The ATA (client / pap2) should register to my switch via a proxy server i.e. SRTP server. I would need setup and training on how to do so in future.

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I am seeking a techie, who could install IP based PBX on my cloud setup. It will be a software based PBX either 3CX or any other asterisk based. Here are the precise requirements:- 1. Install and configure IP based PBX system.(Product shall be decided mutually). 2. Configure TLS/SRTP to traverse voice over ssl. 3. Ensure that traffic is passed through blocked gateways. 4. Configure auto provisioning for IP Phones and mobile clients. 5. Mobile clients must be compatible across various OS. 6. Configure SIP trunk for International calls (preferably google voice). 7. Any other work that shall be required for smooth implementation and functioning.