Hi, I have an application (SLIC/DSP software) on our embedded device (has POTS ports) that handles RTP incoming and outgoing traffic once RTP dest IP/port and local RTP port are provided. I want to extract RTP src/dest port information from SIP messages and pass on to this application during call creation, so that it can manage RTP streams. Till now, I am able to establish SIP calls on our device but audio is not working as pjsip opens RTP socket which I want to open on my RTP application. # netstat -anp | grep 400 udp 0 0 0.0.0.0:4000 0.0.0.0:* 1571/pjsua- mips-unk udp 0 0 0.0.0.0:4001 0.0.0.0:* 1571/pjsua- mips-unk udp 0 6400 Is it possible to do this easily? i.e., to use pjsip only for SIP signalling and disable RTP socket creation in pjmedia. Regards, SMS _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list -- pjsip@xxxxxxxxxxxxxxx To unsubscribe send an email to pjsip-leave@xxxxxxxxxxxxxxx
