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