sanfyduan at gmail.com
Fri May 3 01:55:32 UTC 2019
trunking <---> B(pstn phone).
1. create new sessionB then bridge A to B:
(1) sessionA.answer() and ring back to A,
(2) create new session to B via sip bridge, then break sessionA ringback
and park sessionB,
(3) bridge(sessionA, sessionB)
2. bridge A to B directly without create sessionB:
The second one works well. But in the first one, A always loses some audio
I use tcpdump on my Freeswitch server to check the connection and find that
some UDP packets from Freeswitch to A are lost, but the packets from Twillo
to Freeswitch are not lose.
It seems that some UDP packet are lost while the Freeswitch relays the udp
packets to A.
So, what the difference is between the two methods? And how can I fix this
problem? I need to use the first method to make the call.
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