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SRTP / SDES / TLS-SIPS on asterisk or whatevever SIP Server / Proxy / PBX

I have one Voip Phone APP on Symbian , Based on the SDK pjsip ([url removed, login to view])

I guess the app are fine , i want to work with SRTP , TLS/SIPS and the key excange are the SDES

By the way, i think the APP are fine cause when i connect the app on "[url removed, login to view]' or '[url removed, login to view]' i have good conversation and check in logfiles that´s everything goes fine . But when i want to use in my Asterisk server i have an 'one way audio' problem , in freeswitch some kinda of 'unknown media type' and stuff. So, i can shoot that are problem with my server.

By the Way i need specifcly that´s my server works correct with SRTP-SDES and TLS-SIPS . I have in my server Asterisk and freesitch already Instaled.

In first moment as related above is just an server correct ajust.

Will Be Waiting The Bids


Kemahiran: Linux

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Tentang Majikan:
( 0 ulasan ) Curitiba, Brazil

ID Projek: #362884

Dianugerahkan kepada:


Working in SIP based application sience last 4 yrs. If The Problem is only at the Application side can be resolved in specific days.

$100 USD dalam 4 hari
(0 Ulasan)

3 pekerja bebas membida secara purata $167 untuk pekerjaan ini


Fugenx Technologies is a team of experts dedicated to Web development with updates technologies like Ajax, Ruby on rail, Joomla web 2.0 services. Professionals on php, html, xml, java scripting.

$200 USD dalam 20 hari
(0 Ulasan)

3 years working in Asterisk -based PBX system . We have several solutions for a one way audio problem. Indeed the problem is on your server.

$200 USD dalam 3 hari
(0 Ulasan)