sip - I'm developing RTP server. I'm not get ACK to my OK. What am I doing wrong?

Question

This is the message I INVITE I got.

INVITE sip:1@104.154.78.142 SIP/2.0
Via: SIP/2.0/UDP 31.168.230.133:5060;branch=z9hG4bK2516f06a
Max-Forwards: 70
From: "0555042354" <sip:0523737233@31.168.230.133>;tag=as7352ba2e
To: <sip:1@104.154.78.142>
Contact: <sip:0523737233@31.168.230.133:5060>
Call-ID: 6d1cfdd63265a5593c945e85543b4de7@31.168.230.133:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.18.0
Date: Thu, 04 Feb 2021 18:41:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1968686840 1968686840 IN IP4 31.168.230.133
s=Asterisk PBX 11.18.0
c=IN IP4 31.168.230.133
t=0 0
m=audio 16454 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

This this is my response to UDP 31.168.230.133:5060

SIP/2.0 200 OK
Call-ID: 6d1cfdd63265a5593c945e85543b4de7@31.168.230.133:5060
Via: SIP/2.0/UDP 31.168.230.133:5060;branch=z9hG4bK2516f06a
From: "0555042354" <sip:0523737233@31.168.230.133>;tag=as7352ba2e
To: <sip:1@104.154.78.142>
CSeq: 102 INVITE
Contact: <sip:0523737233@31.168.230.133:5060>
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 807151903 807151903 IN IP4 104.154.78.142
s=Asterisk PBX 11.18.0
c=IN IP4 104.154.78.142
t=0 0
m=audio 16455 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecvSIP/2.0 200 OK
Call-ID: 6d1cfdd63265a5593c945e85543b4de7@31.168.230.133:5060
Via: SIP/2.0/UDP 31.168.230.133:5060;branch=z9hG4bK2516f06a
From: "0555042354" <sip:0523737233@31.168.230.133>;tag=as7352ba2e
To: <sip:1@104.154.78.142>
CSeq: 102 INVITE
Contact: <sip:0523737233@31.168.230.133:5060>
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 807151903 807151903 IN IP4 104.154.78.142
s=Asterisk PBX 11.18.0
c=IN IP4 104.154.78.142
t=0 0
m=audio 16455 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

The call initiate. I'm getting the RTP packets, but I don't get ACK.

Later I trying to hangup, by sending BYE but it doesn't work also.

question from:https://stackoverflow.com/questions/66053000/im-developing-rtp-server-im-not-get-ack-to-my-ok-what-am-i-doing-wrong

Answer 1

Waitting for answers