Hello I am looking for a WebRTC expert. Now we are building Webinar live ...a WebRTC expert. Now we are building Webinar live streaming service. AWS Media Live engine is completed. We have to send stream content([login to view URL]) to RTP Input endpoint of AWS Media Live Service. If you have experience of this, Please contact me. Thanks
...maximum resolution and audio will record at 8khz mono. App should run on Android 4.0. User will select what days and hours of the week to run the video function. Use MPEG if codec if free. UI should be one page only. Use a (pink dot=recording/purple square=stopped)icon in notification bar. Yellow highlighted elements in UI picture are optional extra
i need someone experienced about issabel (formerly elastix) to set up an maintain our vps based sip switches. serving to multiple clients using multiple sip operators,trunk or sip user based.
I need an Avaya IP Office expert who is familiar with SIP Trunk configuration with fax server products. Currently we have the licensing installed for 4 SIP Trunk Connections, but the Avaya IP Office is responding with a SIP "Temporary Error".
The VMS platform project should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [login to view URL]
We have an existing project, where we have developed in jque...plugins into react or searching for new ones. we have few bugs like - The rendering takes time in react and the js is already loaded which makes zoom to not work. we are using RTP pinch zoom jquery plugin We have used these technologies Front end - React Backend - node Api - Graphql
I have third party Linux Server, SIP/VOIP, when an IP phones registered with server and are assigned with extension number like 501,502 etc, now these phones make communication with each other by dialing their number. What I want to do is, remove the IP phone and use Raspberry module, and it should work as phone, and connecting push button switch on
...of work but getting frustrated with what looks like sip/nat issue and that I couldn't find a way to be able to call between local extensions. It's mainly for personal use, I wanted a simple PBX that could handle a small group of sip devices that could call each other and that could use an external SIP gateway. The aim was to have better sound quality
...server and then have all SIP traffic route over the VPN to allow 5060 traffic out of Dubai which blocks SIP on 5060. Install IPSEC on the Ubuntu server (currently in production hosting web app) Configure VPN Ensure only SIP traffic flows over VPN lock down the entire solution so the SIP traffic can only go to one IP address our SIP service This is a remote
...already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers. I have already set up the following: 1. Free PBX & Extension 2. Free PBX routing and dial plan 3. Sip trunk to VoIP Provider 4. Outbound dialing rules 5
I need 4 assemblies to be created in Profile Builder version 3: They are: SIP wall (including Internal and External Panels), Concrete Waffle Slab Floor, Mid-Floor and SIP roof. Each Assembly will be treated as a Milestone (the value will depend on difficulty). Outline of each assembly will be provided, but there is a lot of detailed information (both
...to the user based on tags. Content already seen by the user should not be displayed again. The Right hand side of the page links curated content - Game reviews, game data, RTP etc upvote/downvote/report No content management is need for the curate content, these page will all user the same layout template. All content can be "tagged". Users can
Hello I am looking for a Angular5 expert who is familiar with live streaming. Now we ...built AWS Media Live service and are going to build frontEnd interface. Your job is to insert Webcamera or Screensharing feature for live streaming. You need send streaming to RTP Endpoint of AWS MediaLive Service. If you have experience, Please contact me. Thanks
...purchase. Project Pitch We would need an experienced UX/UI Designer to create an appealing visual mockup for our upcoming new digital tool: the Social Impact Profile (SIP). The SIP will be a user database and part of our existing website ([login to view URL]). With this tool we want to offer our customers the possibility to create their own user account
I have written erotic stories for years. Some in which I have published and others in which I have written for other ...erotic stories for years. Some in which I have published and others in which I have written for other people. I love when I get in the groove and just let my mind wander while I sip my wine and free my mind... I'm sure you will too!
...GeoIP Based, ISP Based ) Connection Anti-Drop on stream failure Easy & powerful Transcoding System fast zapping time Supporting All common streaming protocols (HTTP,RTMP,RTSP,RTP,UDP,MMS) TV Archive & Timeshift MPEG-TS, HLS, RTMP Output Fingerprint Sender RTMP Push Anti-Restreaming Concurrent Connection limitation, ISP & Country Locking Incredible Fast