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.
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