Cyber Mega Phone no sound

Asterisk Asterisk WebRTC
1

I noticed that I got no sound because of these errors. What do i have to change?

[Feb  4 13:57:31] ERROR[32131]: res_rtp_asterisk.c:3507 rtp_allocate_transport: Oh dear... we couldn't allocate a port for RTP instance '0x7fa130327210'
[Feb  4 13:57:31] WARNING[32131]: res_rtp_asterisk.c:2570 dtls_srtp_stop_timeout_timer: Unable to cancel schedule ID 0.  This is probably a bug (res_rtp_asterisk.c: dtls_srtp_stop_timeout_timer, line 2570).
[Feb  4 13:57:31] ERROR[32131]: res_rtp_asterisk.c:2570 dtls_srtp_stop_timeout_timer: Invalid refcount -1 on ao2 object 0x7fa130327210
[Feb  4 13:57:31] ERROR[32131]: res_pjsip_sdp_rtp.c:249 create_rtp: Unable to create RTP instance using RTP engine 'asterisk'
[Feb  4 13:57:31] ERROR[32131]: res_pjsip_session.c:1679 ast_sip_session_refresh: Failed to generate session refresh SDP. Not sending session refresh
       > 0x7fa130318110 -- Strict RTP learning complete - Locking on source address 192.168.1.5:39109
2

Dana[1] is a more up to date and maintained client. I’d suggest trying it.

[1] GitHub - nimbleape/dana-the-stream-gatekeeper: React based front-end demo for Asterisk's SFU

3

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