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This is likely due to the same reason as #665 - the rtpstream play thread gets hopelessly behind and all your audio getsthrown away in the jitter buffer at the receiving end.
Hello,
I try to play an encoded audio file in SRTP but hear nothing.
First of all, I add the following lines in INVITE:
use the following RTP description:
and change transport mode to TLS one socket:
sipp -sf ./call.xml -t l1 -tls_key ./cakey.pem -tls_cert ./cacert.pem -s "$USER_ID" -ap "$USER_PWD" -key number "$NUMBER" -d "$CALL_DURATION" example.com:5061
Calls successfully connect. But I hear nothing.
If I use the pattern example from the rtpcheck_xml_syntax_reference.pdf I hear the clicks.
Are there any restrictions for using audio files in SRTP scenarios?
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