Voip rtp sniffer software jobs

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    7,461 voip rtp sniffer software jobs found, pricing in CAD

    ...Streams & Users (With Authentication) ✔️ Live Streams, VOD ✔️ Easy & powerful Transcoding System ✔️ On-Demand ✔️ Supporting All common streaming protocols (HTTP,RTMP,RTSP,RTP,UDP,MMS) ✔️ TV Archive & Timeshift ✔️ MPEG-TS, HLS, RTMP Output ✔️ Create your Own Live Channels by combining multiple videos ✔️ Powerful Logging, Statistics & Connection

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    We are a Telecom company based In Hong Kong, We are Looking for Interested candidates with Knowledge in VoIP and call center Solutions, who can Re-Sell our Services to call center and Earn Huge Incentives on Every Invoice Raised for the customer.

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    Hi, Need to redesign launcher icon and other screens for voip mobile dialer app. Name of app is orbitcall you can find it in play store and itunes store. Please come up with some sample launcher icons for the app and the cost and time estimate. Thanks Vidhya

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    I need to sync rtp with caller and called with RTCP SR. Project must be done in c# . SSRC and Payload changes. Codecs are G711,G729,AMR WB,AMR NB,EVS.

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    want a voip broad cast software using voip. cli routes we can buy please mail [removed by freelancer.com admin[for details

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    We would like you to help us to make a Java project who was able to do a call throught a VOIP channel and play a text (text2Speech) and recognise DTMF tones that the person would do in his keyboard. For example: Hello Mr John Smith, press 12 if you have received the package otherwise press 49. Can you help us?

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    Hi, I need someone to get my excel list of 200 online slots games and fill out the blanks with the following data for each game: Paylines Reels Progressive Software Theme Tags (max. 3) RTP Data will be verified by operators, so everything needs to be as accurate as it gets.

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    ...program without interface, which can connect to a PBX using VoIP SIP, the most important is to connect using TLS transport. I don't have the problem if you can create the program from scratch or using Ozeki SDK, no any other choice accepted. The result should be the same as the other software that is exist, I've already created a program using the documentation

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    INFORMATION: We are using Freeswitch as Backend server and Customized Linphone app and both It is working fine together. REQUIREMENTS: Now we want to integrate Flexisip Server as proxy server to handle Push Notifications when device is offline. Installation: [login to view URL] Configuration: [login to view URL] Modules: [login to view URL] Push Gateway: [login to view URL] Flexisip on Github: ...

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    Details will be shared winning bidder. please bid if you have the experience.

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    hi, am looking to build a voip calling service. if someone can help me with this..please lemme know

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    voip Ended

    USB dongle with raspberry to work as gateway setting

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    @ need one app which should be installed in android and ios device @ the app should be able to contact other app via any medium @ call clarity to be clear @ video call also should be possible @ the end to end communication should be such that it should not be identified by any service provider as they block voice packets

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    Hey there i need US virtual numbers that can be used to verify [login to view URL],you have to check that your numbers can work for that site..

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    I need VoIP Softphone With Sip Unblocking Proxy. If you have any solution please contact me.. I need only android Version..

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    We are looking for VoIP expert to join our noc team for monitoring and analysis of our current routes, you must be have previous experience. You will be dealing with VoIP routes, identifiying bad numbers on the routes using the CDR and Call filter, adjusting the dynamic call filter and Human behaviour server. Other tasks will be discussed in more

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    VoIP service provider has a project for a programmer to create a softphone from existing open source code. The end result will be a working Softphone running on Android OS and meet license "GNU General Public License, version 2" at [login to view URL] Must be able to work with existing source primarily Java and XML and modify to

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    I would like to have some kind of voip integration to be able to know what client is calling, what are the future meetings etc, and be able to create new client from this etc. I do not have this crm. :( but the basic features I would like [login to view URL]

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    Hello, I'm looking who can make a video player send a custom user agent using php, pyton, curl or any...</body> Important: the user agent can't be in the url for example: [login to view URL]:User-Agent:1111 Using that method can be track down using a simple URL sniffer. So I'm looking a method that would not be that simple to trace.

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    ... Supported Codecs + MPEG2/MPEG-AVC/H264 + AAC/MP3 Streaming Protocol + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS + HLS SERVER Management + Master / Slave Server + Act as a Master Server, a Slave server or both + HTTP protocol between Master and Slave Server VOD + Ability

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    I would like to offer my customers VOIP plans using Twilio as my wholesale VoIP provider. Would you be able to create a tutorial for me on how to A. Create new numbers and get the SIP configuration for the VOIP ATA or VOIP enabled wifi modem/router. and B. Port in a number and get the configuration for the VOIP ATA. I need a step by step guide in

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    ...would like to offer my customers VOIP plans in Australia using Twilio as my wholesale VoIP provider. Would you be able to create a tutorial for me on how to A. Create new numbers and get the SIP configuration for the VOIP ATA or VOIP enabled wifi modem/router. and B. Port in a number and get the configuration for the VOIP ATA. I need a step by step guide

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    We are customizing fusion pbx with a billing system together. This has been done successfully although untested. Now few things remaining on the project: 1. We want to add API for international mobile top up or mobile recharge - this will involve adding security and other things that can stop frauds from happening. We will present a guide to the successful developer. We also need reports for a...

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    Hi, We have portsip voip sdk and need to develop user interface based on that sdk. Following needs to be done for both android and IOS. 1. Splash Screen -show image/animated image 2. Login Screen -enter ip, port, user , pass , advance options 4. Dialpad Screen -show Number pad to dial numbers 5. Recent Calls Screen -show dialed, answered,missed

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    I need an Android app. I would like it designed and built. Voip dialer .. with encription

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    ...develope a software for me. I need to understand how to convert an hi3 payload to audio AMR format, and listen to it. The file is made by some stream with TCP header, UDP header, RTP header and rtp payload. Any crypto algo is applied. i have the file and i need to UNDERSTAND how to create an amr audio file from that stream. the software will be

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    the software will include below features: text audio video chat screen sharing contact list admin panel with quality control features, call recordings, screen watching, call joining, call forwarding and some other features mentioned in documents attached to this project. total cost of this project and other related information are within

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    I need someone to help configuring GoIP-1 as a trunk on an 3CX voip server, for both outgoing and incoming calls. Here is the device info and manual, but the current firmware differs from the manual. ( [login to view URL])

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    I want to produce a web app that allows call divert. This will be from a simple api in jason format Geographical number > Mobile number If intrested please pm for further info

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    ...under load Streaming Protocol: + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS Live Encoding Protocol: + UDP Multicast + RTP Multicast + RTP Unicast + UDP Unicast Player Management: + Periodically Check Players' status + Remotely Reboot Player + Remotely Configure Player Setting

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    Circle CI 2 configuration file that uses Docker to install Magento2/php/mysql/phpunit. Looking for someone to setup magento 2 CI env to run code sniffer, tests etc on Circleci. Including static, phpunit and integration test as well as jasmine for javascript etc as a circleci config. Circle Ci will be linked to Bitbucket.

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    I am looking a voip dialer which work in voip block countries. if some done before please contact me

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    ...net connection, we can serve them a streaming video ad specific to LA natives without worry of a bad user experience because of a slow connection. So we'd like to create a sniffer code to be added to our ads that would detect what type of connection a user has and their IP address for location. That information will then be collected to a database. Just

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    ...****************** + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS Live Encoding Protocol ********************** + UDP Multicast + RTP Multicast + RTP Unicast + UDP Unicast + Recorded TV on Demand + Record TV programs to archive + Master / Slave Server + Act as

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    Hi, I'm currently setting up a hosted VoIP phone provider company for businesses using an Asterisk server. I'm looking to set up the cheapest option for DID numbers, inbound SIP trunks, outbound SIP trunks, and SMS. There are a lot of ready solutions but it's very expensive. Does anyone know how I can set up my own SMS server and my own SIP trunks

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    Preciso montar um servidor em nuvem para montar uma central Asterisk Voip IP

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    Need someone to be very strong with DNS setup understanding. 1. Setup account in WHM 2. Setup ALL DNS settings to point correctly. A records RTP records 3. Configure our privet name servers ns1.arcaveroom.com. 4. Make sure the hostname is setup correctly in WHM 5. After you setup the DNS i must use a DNS checker to make sure the site points correctly

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    Hi, i need help of a Windows engineer who is proefficient with VOIP and Linux. I will share details with the selected candidate

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    Es soll ein voip account erstellt werden, der gegen Spam calls und call id spoofing geschützt ist

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    Linux Script voip Ended
    VERIFIED

    I need a script that calls from linux , i need to setup how many lines to open , what exit i want to give , and what number i want and how many minutes to stay open

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    ...flagship product is a VoIP app for App to App voice, video calling as well as chatting, App to phone calling as well as phone to phone calling using DID numbers on the apps and we we chosen to work with Sinch team. We need a developer that can coordinate with sinch to complete our project work so we can be ready to go live on our VoIP. Kindly advice

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    I am looking for someone who can take care of this project as soon as possible check the attached file

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    We need complete solution to run our gateway for voip termination. We have GOIP-32 port gateways and we need to run on puppy linux. So first we need to compile puppy linux which can be installed on PC and then all gateways will be connected to PC through switch or hub. web based panel we need for dial plan and routing. where we can see live calls

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    Platform iOS: We require a component that can dial a phone number within the app, meaning not going out of the app so we can control when to hang up the call and to what number will be dial.

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    I am looking for developer that can develop a python packet sniffer and packet injection tools.

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    I want an application that: 1- Should collect data from 3 different websites; (each website contains 3 to 8 links) 2- A refresher in the application. 3- Skype or VoIP calling system.

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    We are looking for experienced content writer for websites for services voip, call center setup, telecom, ippbx.

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    ...****************** + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS Live Encoding Protocol ********************** + UDP Multicast + RTP Multicast + RTP Unicast + UDP Unicast + Recorded TV on Demand + Record TV programs to archive + Master / Slave Server + Act as

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    install the pbx software

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