Asterisk VoIP Gateway interfacing to GSM network AMBE+2 Codec
$5000-25000 USD
Закрито
Опублікований over 13 years ago
$5000-25000 USD
Оплачується при отриманні
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