Asterisk VoIP Gateway interfacing to GSM network AMBE+2 Codec
$5000-25000 USD
Closed
Posted over 13 years ago
$5000-25000 USD
Paid on delivery
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
Project ID: 3809958
About the project
2 proposals
Remote project
Active 13 yrs ago
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About the client
United States
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Member since Oct 23, 2010
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