I can see that the sip phone is sending the RTP but the webserver fails to send as seen from the channelstats in asterisk
2 active SIP channels
ip-172-31-24-111*CLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
174.62.206.166 65cda1c11e0 00:01:10 0000002546 0000000403 (13.67%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0073
34.204.51.242 44323dc0-26 00:01:10 0000000000 0000000001 (100.00%) 0.0000 0000002671 0000000000 ( 0.00%) 0.0000
2 active SIP channels
ip-172-31-24-111*CLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
174.62.206.166 65cda1c11e0 00:01:10 0000002566 0000000403 (13.57%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0073
34.204.51.242 44323dc0-26 00:01:11 0000000000 0000000001 (100.00%) 0.0000 0000002691 0000000000 ( 0.00%) 0.0000
Can you please help on this?Thanks for your help..