25 Oct
2019
25 Oct
'19
2:46 p.m.
Hi all,
No Audio flows between the two parties when one side resumes the call after
putting on hold.
Configuration - WebRTC <--> Kamailio <--> Asterisk
Below is the warning I'm getting in asterisk console
WARNING[32388]: chan_sip.c:10425 process_sdp: Declining non-primary audio
stream: audio 10596 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112
113 101
Is this a codec issue or something else.
Thanks,
Arish