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Sip Flash Phone |
| Status |
Closed |
| Budget |
$1000-3000 |
| Posted |
24 Apr 2008 05:47 AM |
| Expires |
24 May 2008 05:47 AM |
| Posted By |
flashphone |
| Service buyer Rating |
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| Description |
Target is to programme a flash based VoIP (SIP) Phone / Server like flashphone.ru, tringme.com or ribitt.com. Following are the requirements:
- Implementation of a server side framework to allow flash clients to place and receive phone calls from / to the PSTN like flashphone.ru or tringme.com or ribitt.com
- Server side application that is based / or connected on / with the Red5 Java Flash Media Server
- On the server 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. So the RTMP stream from flash has to be converted to the RTP VoIP stream.
- 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 5060, and a corresponding audio stream (RTP stream) that uses higher ports between 30000 - 60000
- The process also has to work the other way around. If an audio VoIP call is received on the Asterisk VoIP server it is send to the red5 server that has to convert the call to a flash media call.
- The implementation should be based on the following open source components: Red5 (Java Flash media server), Java SIP Protocol stack (e.g. Jain), Open Source RTP stacks for Java (just search for RTP/java in google), There is also a small C application available that converts the nelly moser coded to G711 (e.g. http://code.google.com/p/nelly2pcm/)
The job is basically, to use the existing open source libraries, integrate them and make them work together.
Please bid only if you could fulfill the requirements and if you are experienced in VoIP.
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