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JsSIP is currently trying to connect to audio (after accepting incoming call) to random ports over UDP. How can I restrict it to a specific to a port range for the audio?

Edit: Log from Asterisk Exp:m=audio 63485 UDP/TLS/RTP/SAVPF 8 107 0 101 (here we got port 63485 which is out of range )

Sanket Pandia
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1 Answers1

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Asterisk port range for media should be configured in rtp.conf

default:

rtpstart=10000
rtpend=20000
Pierre Noyelle
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