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    1,471 rtp jobs found, pricing in USD

    Hi, I have installed the wonza media server in windows server 2003 it looks working fine, then I use the encoder to start my streaming and it the connection looks ok . I use server: rtmp://[wowza-address]:1935/live steeam: and I get connection to the server. But when I want to test the streaming url: rtsp://[wowza-address]:1935/liv...doesn't work. What I am trying to do is: start a radio streaming using wonza media server to hear my streaming in an android phone. this is a sample of streaming from other guy which is working in android phones: rtsp:// I try the tutorial from this site: Use as refence, thank you.

    $30 (Avg Bid)
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    1 bids

    Dears , We are looking to have Webdialer Flash based , Sip Softphone , with possible Encryption solution embeded So that can wo...support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. More detail will be provided at start of contract Regards Forhan Fizan Telecom

    $2933 (Avg Bid)
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    6 bids

    What is required: A client-server type application written in C++ for streaming audio across a network. Server requirements: Must use RTP/RTSP protocol, have a playlist for dragging-n-dropping files to stream, and support upto 100 users. Client requirements: Must use the RTP/RTSP protocol and basically get the audio from the server. Other: Code must be as SIMPLE as possible and preferably include comments; any design documentation must be supplied and complete confidentiality MUST be assured (files REMOVED from your system/any online storage after delivery).

    $141 (Avg Bid)
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    3 bids

    Dears , We are looking to have Webdialer Sip Softphone ,Mobile Dialer symbian with possible Encryption solution embeded So that can work from Blocked areas . Required features: 1. Based on SIP. 2. Codec G.7...silence supresseion. 3. Show Balance inquiry, real time balance display,Maximum call duration,Destination,cdr call history 4. DTMF support. 5. Traverse NAT/Firewall/Blocked network through Encryption VPN or Tunneling or any 6. SoftPhones to communicate with one encryption server for both media and signaling. 7. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 8. The whole system should work with standard softswitch. 9. Registration of softphone clients through the encryption server in the softswitch. T...

    $4500 (Avg Bid)
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    2 bids

    Hi! I need a web based JAVA-SIP client to add SIP phone capabilities to web applications with a Javascript API . Something like The idea is to host the "core" jar files in our media servers: our customers can open a popup (link) in their websites that will call our domain () in order to use the webphone. Requirments: * SIP and RTP stack (compatible with standard VOIP servers) * Standard java applet (no installation required. runs directly from browsers) * Should work on Windows, Linux and Mac and support the most popular web-browsers: Firefox, IE, Chrome, Opera. * Fully skin-able (HTTP/CSS). Should be easily modified (by ourselves) to match the look and feel of existing web applications. *Standard G711 codec’s (PCMU and PCMA) * SIP

    $1200 (Avg Bid)
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    I'm looking for simple RTSP client (in java) which is able manage RTSP handshake, receive RTP stream, decode H264 and export images as they come in. The video server support only authenticated RTSP session. So my vision is to edit some properties file (url, user, pass etc.) and start the program which will connect to RTSP server, receive RTP stream, decode H264 and save images to output directory. There should be posibility to manage framerate - let say the server is sending 24 images per second and I can manage them all. But sometimes I don't need 24 images but 5 is enough. Program have to be in java and can use other open source java libraries.

    $522 (Avg Bid)
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    9 bids

    ...start sending stream of data via RTP (over UDP). The software will receive stream, decode, and from video recognize the kart number. There can be 0-10 karts on the video in one time, most often there is none or one kart on the video. Kart numbers can be 1,2,3..20. If the kart (kart number) is recognized it will just write the kart number and current time to the console (later it will call existing method). By default, everything should happens on the background without video player on the screen, but for human eye check, it have to be able to show player in a very simple window (swing frame) and mark the detected kart number by rectangle (object tracking). So, main features are: - connect to video server and start RTSP session with authentication - receive RTP stream ...

    $1210 (Avg Bid)
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    Dears , We are looking to have Webdialer Flash based , Sip Softphone , with possible Encryption solution embeded So that can work from Blocked areas . Requir...support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. More detail will be provided at start of contract Regar...

    $2666 (Avg Bid)
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    3 bids

    Dears , We are looking to have Webdialer Flash based , Sip Softphone , with possible Encryption solution embeded So that can work from Blocked areas . Requir...support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. More detail will be provided at start of contract Regar...

    $5643 (Avg Bid)
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    7 bids

    We need to create opensource SIP server solution with SRTP/TLS. The overall solution needs to support TLS with SRTP connection with Cisco WIP310-G2 phones. I will only accept bidders that have experience in TLS - SRTP and have SUCCESFULLY implemented this in the past. When bidding, please explain the solution in detail (how you have already implemented this) and the time to delivery.

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    We need to create opensource SIP server solution with SRTP/TLS. The overall solution needs to support TLS with SRTP connection with Cisco WIP310-G2 phones. I will only accept bidders that have experience in TLS - SRTP and have SUCCESFULLY implemented this in the past. When bidding, please explain the solution in detail (how you have already implemented this) and the time to delivery.

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    ...start sending stream of data via RTP (over UDP). The software will receive stream, decode, and from video recognize the kart number. There can be 0-10 karts on the video in one time, most often there is none or one kart on the video. Kart numbers can be 1,2,3..20. If the kart (kart number) is recognized it will just write the kart number and current time to the console (later it will call existing method). By default, everything should happens on the background without video player on the screen, but for human eye check, it have to be able to show player in a very simple window (swing frame) and mark the detected kart number by rectangle (object tracking). So, main features are: - connect to video server and start RTSP session with authentication - receive RTP stream ...

    $1368 (Avg Bid)
    $1368 Avg Bid
    4 bids

    We need to create opensource SIP server solution with SRTP/TLS. The overall solution needs to support TLS with SRTP connection with Cisco WIP310-G2 phones. I will only accept bidders that have experience in TLS - SRTP and have SUCCESFULLY implemented this in the past. When bidding, please explain the solution in detail (how you have already implemented this) and the time to delivery.

    $428 (Avg Bid)
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    3 bids

    We need to create opensource SIP server solution with SRTP/TLS. The overall solution needs to support TLS with SRTP connection with Cisco WIP310-G2 phones. I will only accept bidders that have experience in TLS - SRTP and have SUCCESFULLY implemented this in the past. When bidding, please explain the solution in detail (how you have already implemented this) and the time to delivery.

    $30 - $250
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    I need a client application, that will play stream in MPEG4/H.264 format from a RTSP server. server is basically a network camera, providing live mpeg4 (or H.264) stream via: RTP RTSP and RTSP over HTTP the client should be written in .NET c#. ## Deliverables 1) All deliverables will be considered "work made for hire" under U.S. Copyright law. Employer will receive exclusive and complete copyrights to all work purchased. (No 3rd party components unless all copyright ramifications are explained AND AGREED TO by the employer on the site per the worker's Worker Legal Agreement). ## Platform Microsoft .NET framework 2.0

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    ...following things will now be possible: - Live video streaming View from mobile is possible by dialing a phone number and conference code from mobile smart phones including symbian, java, android, iphone, windows mobile and blackberry without any download and install of application and maintaining the qualities of existing video streaming application and complying required RFCs like RTSP, HTTP, SIP, RTP, and UDP among others. - Accessing the live video stream from website should be continue to be possible like the way it is right now. Its not a rocket science...many folks like o#ctro have already achieved it. It should be fun for folks who love challenges. Do let me know if any queries. ## Deliverables 1) All deliverables will be considered "work made for hire" under ...

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    Looking to develop an Android SIP VoIP client that allows users to use prepaid voice minutes purchased through us. Application should provide the following features (refer to Acrobits Groundwire for reference): Features: • Multitaski...contacts directly from the softphone • Quickdial, your 12 favorite contacts are one touch away • ability to generate DTMF tones while in call, to control various PBX features or automated systems (use • audio, rfc 2833 or SIP INFO) • speakerphone support • detailed call history, with intelligent call grouping for an easy overview • support for sip:username URLs in phonebook • configurable RTP port range • SIP Proxy support, VPN support • STUN server support • Call Waiting and Con...

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    ...new contacts directly from the softphone o Quickdial, your 12 favorite contacts are one touch away o ability to generate DTMF tones while in call, to control various PBX features or automated systems (use audio, rfc 2833 or SIP INFO) o speakerphone support o detailed call history, with intelligent call grouping for an easy overview o support for sip:username URLs in phonebook o configurable RTP port range o SIP Proxy support, VPN support o STUN server support, automatic service discovery using DNS SRV queries o phone number resolution. We present the phone numbers in a convenient format with grouped digits, display the flag and country name and even region/city name for some countries, including USA...

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    ...with support of rtp and udp. supports one to one calling and other SIP features. 2 - vc++ based h.264/gsm/http live video streaming web application (broadcasting client, media server and viewer) with media server written in c#. Works flawlessly to support 10 live streams and 100 viewers in each streams. To do: Merge the two applications and enhance the merged application such that following things will now be possible: - Live video streaming View from mobile is possible by dialing a phone number and session code from mobile smart phones including symbian, java, android, iphone, windows mobile and blackberry without any download and install of application and maintaining the qualities of existing video streaming application and complying required RFCs like RTSP, HTTP, SIP, RTP...

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    Develop a VoIP Gateway interfacing to SIP Voice and SMS traffic, transcode AMBE+2 Codec, and perform RTP header compression. Must be familiar with Session Border Controller to resolve VoIP and SMS traffic from NATed environments. Required familiarity with the following protocols: - SIP - RTP - NAT / IP-Sharing - GSM / AMR Codec/ IuH

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    Dears , We are looking to have Webdialer Flash based , Sip Softphone , with possible Encryption solution embeded So that can work from Blocked areas . Requir...support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. More detail will be provided at start of contract Regar...

    $2580 (Avg Bid)
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    I have multiple single program transport streams. I'm looking for a software solution to create ONE multiplexed transport stream that contain all the streams. The sources are in UDP or RTP format. The streams are LIVE, not file based.

    $250 - $750
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    Need a mobile SoftPhone with source code. Required features: 1. Based on SIP. 2. Should support Voice and SMS. 3. Codec G.72...support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. 11. Wants to use in VoipSwitch like a pc2phone More detail will be provided at start of...

    $2018 (Avg Bid)
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    Requirements : 1) A simple server which puts together the video chunks of a mp4/3gp...amount & plays) 3) Date of Completion : within a week but the it can be extended to few days if required. I would like the project to be completed within the deadline so that I will have some time to test and get back to you for any queries/issues. Notes : 1) Development in eclipse 2) The programming needs to be done at the socket level. 3) Usage of http (URLConnection), RTSP and RTP is strict no. 4) It would be preferred if the video file could be played without downloading it onto the client. 5) Let me know before you start whether you would be using tcp or udp and why. 6) As and when you move on with the project, snapshots of the implementation will be good. 7) A ...

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    This Project aims at building a cost effective multimedia conferencing solution where all the clients of ad-hoc wireless network can easily communicate via video conferencing and also enable text chat, audio chat and file transfer. This project is a software package for making multimedia conference by using java media framework, swings for graphical user interface applications, jdbc and MS-Access database for information storage. The system is built using Wi-Fi wireless network as the medium of communication and will use live video capturing and streaming the way the live TV works but here the difference is there will be two ways communication.

    $487 (Avg Bid)
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    Dears , We are looking to have Webdialer Flash based , Sip Softphone , with possible Encryption solution embeded So that can work from Blocked areas . Requir...support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. More detail will be provided at start of contract Regar...

    $2800 (Avg Bid)
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    7 bids

    Description Need a mobile SoftPhone with source code. Required features: 1. Based on SIP. 2. Should support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion...support Voice and SMS. 3. Codec G.723, G.729 with silence supresseion. 4. Balance inquiry, real time balance display. 5. DTMF support. 6. Traverse NAT/Firewall/Blocked network through Encryption (VPN or Tunneling or P2P...any) 7. SoftPhones to communicate with one encryption server for both media and signaling. 8. Encryption Server will decrypt the encrypted traffic and use SIP for signaling and RTP for Voice. 9. The whole system should work with standard softswitch. 10. Registration of softphone clients through the encryption server in the softswitch. More detail will be provided at start of...

    $3777 (Avg Bid)
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    Description: Provide a VoIP Android library as a software package for embedding into an application Code-name: VoIP Android library (engine) Target platform: Android (1.5 or above) Delivery form: pre-compiled library or sources Requirements: • Supports SIP RFC standards • RTP/RTPC(UDP), an effective jitter buffer implementation • G.711 (A-Law and u-Law), iLBC and optional G.729A/B • Echo Cancellation support, Tone Generator, File playback • Call recording (optional) • STUN support for NAT/firewall traversal (optional) • Supports multiple SIP profiles (optional) • Simultaneous outbound calls and conferences supporting • Ability to configure the cellular ...

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    ...PSTN fail-over life lines. Peripheral Ports: USB, Audio In, Audio Out. Conference Rooms: 2. Unified Message Storage: 75 hours of voicemail, 5000 fax pages, 2 hours of video mail. Registered Extensions: 100. Voice Codecs: G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay. Video Codecs: H.264, H.263/H.263+. Communication/Security Protocols: TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP Compliance: FCC: Part 68 & 15B. CE: EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 A-Tick : AS-ACIF S002, AS/NZS60950 UL (power supply) Universal Power Supply: Inp...

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    a single common mobile software installed in any mobile phones with Java MIDI2.0 supported to make call out and receiving incoming calls. Software must be developed in J2ME and support all kind of mobile OS including Symbian, Win Mobile, Blackberry, Maemo (Diablo, MAemo, Meego), Android, ...incoming calls. Software must be developed in J2ME and support all kind of mobile OS including Symbian, Win Mobile, Blackberry, Maemo (Diablo, MAemo, Meego), Android, Bada and embedded OS (RTOS) devided into 2 parts: 1. post dial dtmf send dtmf mode to Telephony Server including pin code, 1 or 2 parameters and destination phone number 2. voip call call out using SIP/RTP to SIP Server, received SIP/RTP call from SIP Server source code included, job required now and to delive...

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    We need to develop a Windows based application that will capture Cisco SKINNY call control information and the corresponding RTP traffic, and reconstruct it in to audo files. ## Deliverables we plan to use WinPcap to capture VoIP traffic for this application.? Knowledge of Wireshark is important.

    $500 - $5000
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    ...service however with only four servers. The first server will play the role of a sip proxy, load balancer, SBC and media proxy. The other two will be telephony servers (i.e asterisk) that will handle the calls. The system must be scalable so when the capacity is reached on the servers, a more can be added. You are required to utilize: -A good SIP proxy to take care NAT bursting and media RTP issues (OpenSips?Kamailio?Freeswitch? You need to select which one and justify the reason behind.) -Freeside to handle the billing aspect Here you are required to enable the following common functions: 1)Voicemail: Users should be able to see (and listen) to their voicemails via their freeside Customer interface. 2) A follow me number. Users should be able to select a follow-m...

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    SIP softphone with encryption on SIP packets and smooth on RTP packets. Contact me asap for more information. Only who have done similar project need to contact.

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    ...For example, a university might have such a server, which redirects requests for PERSON-X-EMAIL to departmental servers. Explain how a location request for PERSON-Y-EMAIL might be redirected to PERSON-Y-ALTERNATE-EMAIL Question 3 : does RTSP differ from HTTP (a) in maintaining client state information; (b) in terms of in-band, out-of-band traffic? RTP RFC and find out how to calculate the inter-arrival time jitter in the RTCP reception report. Question 4 : Two hosts want to communicate with each other. Each host is protected by a firewall that filter packets based on packet header. Explain the rationale for having an AH tunnel between gateways and using ESP in transport mode between the two hosts. Question 5 : The CEO of a me...

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    I need a? WEB based application that must run on iPhone and captures the audio and video stream from iPhone's camera and mic., encodes the captured stream using H.264/AAC encoding format and sends the encoded stream over Internet to a streaming server using RTP protocol. Is it possible to build such a solution? Which technologies can be used?? Please BID only you have previous experience with iPhone apps. and you have a viable solution. Coders with the best solution will have an advantage.

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    We need a full skill Software engenier, avaible to reproduce linux IPTV Streaming server ready for DVBS2,ATSC,DVBC, ect,ect. most have billing solution , addon package for content. Most be a streaming server designed for implementation in IPTV, Video-on-Demand, Audio-on-Deman...Billing solution Key Features * High-Quality MPEG4 and H.264 Video Streaming * High Streaming Capacity * Stream Redistribution Support * TCP Streaming Support * UDP Streaming Support * Multicast Streaming Support * Subscriber Authorization and Stream Encryption * Web-Based Remote Management * Fully Integrated IPTV Delivery Solution * RTSP, TS, RTP, HTTP,ASF, AVI, MMS Http,mvk Streaming Support * 200+ Over-The-Top (OTT) and Free-To-Air (FTA) Chann...

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    Here is problem definition as you have requested. We have developed peer to peer audio chat application for PC. It uses RTP protocol built over UDP. We are using port range from 5000-6100. Application works perfectly when two machines are in same LAN and also when two machines are directly connected to internet. There is a problem which we are facing while working in following case: When one machine is behind router and in LAN(Domain Controller) and when other machine is in other network and directly connected to internet. Then the machine which is in LAN and behind router not getting feeds of other machine. But other machine which is connected directly to internet is getting feeds of machine in LAN. We need to understand and need help on the following a.) How do w...

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    ...is that the source some times stops, and the encoding stops as well, but my encoding executable still thinks that its running. It doesn't stop, BUT, the CPU drops to 0%. Either fix it that it doesn't ever stall and drop to 0% and or add a watch service that can detect the process has stopped or 0% cpu and restart it. #2 issue is that I need to receive the stream in rtp:// format. Right now I am receiving my source in rtp:// and converting it to http:// with vlc. Either find or create a direct show filter that can take this or embed vlc into the application so that it is invisible and use vlc to receiving the stream and change it over to http:// I also want to be able to set any input streaming format as a source. Udp rtsp and in any format h.264, flash, an...

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    Need an OS X verson of a Windows-based audio client.? Client should run in the background with a seperate GUI for configuration.? Background client application? should receive MP3 or WAV audio via TCP or UDP and play audio through the sound card. ## Deliverables I need a ...from server and stores it in the file system.? ? After file in on system, plays MP3 or WAV audio file to sound card.? The messaging is very simple, but we can provide the protocol for the messaging. UDP MP3 Audio - Opens a? UDP port? a listens for MP3 data sent by raw UDP.? Performs some basic buffering of MP3 data and plays MP3 audio to sound card. UDP RTP uLaw - Opens a UDP port and listens for uLaw audio data packets with RTP headers.? Performs basic buffering of uLaw data and play uLaw aud...

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    We have our own VOIP recording system which use mirror/sniffing on the switches to recorded the calls. We are depending on a hardware PCI card to resolve the signaling and forward the RTP of the calls to an IP address. I’m using my own service to capture all the RTP and record it. I need to replace the hardware card I’m using and recording using a different way than sniffing. I need a service to make the following with Avaya PBX: Handling of TSAPI messages in the areas of event notification and value queries to monitor a specific extension. Use of Avaya AES DMCC registration services to register and un-register the virtual IP softphones. Use of Avaya AES DMCC call control services to activate Single Step Conference for the virtual IP softphones. ...

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    We have our own VOIP recording system which use mirror/sniffing on the switches to recorded the calls. We are depending on a hardware PCI card to resolve the signaling and forward the RTP of the calls to an IP address. I’m using my own service to capture all the RTP and record it. I need to replace the hardware card I’m using and recording using a different way than sniffing. I need a service to make the following with Avaya PBX:? ? ? Handling of TSAPI messages in the areas of event notification and value queries to monitor a specific extension. ? Use of Avaya AES DMCC registration services to register and un-register the virtual IP softphones. ? Use of Avaya AES DMCC call control services to activate Single Step Conference for the virtual IP sof...

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    ...side the audio stream from an standard audio flash client (already exists) has to be converted from the Nelly Moser Codec to G711 audio codec in real-time or use Speex above Flash Player 10. So the RTMP stream from flash has to be converted to the RTP VoIP stream. For video can use h263,h263p,h264. - In addition to that the corresponding SIP signalling to a VoIP application server has to be established. Explanation: The VoIP protocol consists of a SIP signalling text based protocol which used port usually 5060, and a corresponding audio stream (RTP stream) that uses higher ports between 30000 - 60000 see: - The process also has to work the other way around. If an audio VoIP call is received on the VoIP server it is send to the red5 server that has to convert the call t...

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    1) I have an IP camera(Q1755). that deliver the streaming video from it. this camera can stream a video by rtp/rtsp/http tunnel 2) I want a source filter that connect to this camera and streaming this tunnel. 3) after that I want a transform filter that can extract rtp from this tunnel and save this streaming (rtp) into a file that can be displayed later by any video player 4) make this code by visual studio c++ and connect between these filters to be one program that extract rtp from rtp/rtsp/http tunnel and save it by render filter to be displayed later 5) use visual studio c++ with windows SDK (directshow) ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) D...

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    ...Video * The Codecs H.263 and H.264 must be supported * Audio * The Codecs GSM and mp3 must be supported * Streaming Server * the solution should work together with the Darwin Streaming Server * open to other solutions * Transmission * send and receive of streaming data: * via RTP over UDP * via RTP in TCP interleaved mode * Proxy-Passthrough * Small Bandwith Mode (128 KBit/s Upload) * Security * Stream-Encryption * How can a stream encryption be applied? Using Bit-Scrambling ? (a Secure-Key exchanged via a secure channel between the participants is applied using XOR with the stre...

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    $2812 Avg Bid
    9 bids

    ...immediate access to leave information based on his CLI recognised by the account set up. if a caller dials in to the mailbox, he will hear the message, if he chooses to dial the key for more information, the call is diverted to the set up client number and the client is billed by the system for the call. You are required to utilise: -A good SIP proxy to take care NAT bursting and media RTP issues (OpenSips?Kamailio?Freeswitch? You need to select which one and justify the reason behind.) billing engine for monthly fees and onward calling Our clients should be able to select a follow-me number that will be billed according to their active tariff. (If the forward is to a voip (SIP) number then there's no charge. ------------ I will provide you with access to ...

    $750 (Avg Bid)
    $750 Avg Bid
    1 bids

    Project is based on opnet simulation. example of topics to simulate(ANY ONE OF THEM): gigabit ethernet ip-multicasting voice over ip prootocola and alogrithms mac layer protocols(sensor networks,ad-hoc networks) routing protocols(simulation of AODV or DSR etc) TCP, RTP , SCTP voip performance ,SIP protocol use of opnet simulation : define your problem, state your assmption,show your design parameter,result and performane parameter,graphs, and conclusion ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows (depending on the nature of the deliverables): a) For web sites or other server-s...

    $30 - $5000
    $30 - $5000
    0 bids

    ...only two servers (or 3 max). The first server will play the role of a sip proxy, load balancer, SBC and media proxy. The other two will be telephony servers (i.e asterisk) that will handle the calls. The system must be scalable so when the capacity is reached on the fist two servers, a third can be added. You are required to utilise: -A good SIP proxy to take care NAT bursting and media RTP issues (OpenSips?Kamailio?Freeswitch? You need to select which one and justify the reason behind.) -A2billing for the billing part. Here you are required to enable the following common functions: 1)Voicemail: Users should be able to see (and listen) to their voicemails via their A2Billing Customer interface. They should also be able to listen to their voicemails via their sip p...

    N/A
    N/A
    0 bids