I have done call forwarding in my mobile phone (say Entity B) and the call gets forwarded to asterisk.
Entity A makes call to Entity B which forwards the call to Asterisk
Asterisk play some music file and Dial to Entity C (webRTC).
until C answers the call, A can hear the Music files played by Asterisk and the ringing from C but as soon as C answers the call there is a missing voice at both ends A and C for about 6 to 7 seconds.
after wireshark analysis i found that Asterisk changes the SSRC for RTP stream from Asterisk to A which takes 6 to 7 seconds and there is no RTP packets for that duration of SSRC change.
my question is why asterisk changes the SSRC for RTP stream after the call gets accepted.
Also if i directly make a call to Asterisk without forwarding than there is no missing voice in this case.