Peer to peer communication through public ip

Dear All,

I have asterisk server and i used a modem for port forwarding now i got my server in public ip
I registered to sip peers in the server by using domain as public ip through internet…
I try to dial from 100 to 202…both are softphone registered in my asterisk server…
used domain as my asterisk server public ip…

But if trying to call a sip to sip…ie…100 to 202 peer …the call originated and answerer but voice is not getting…

Please help me

my last sip packets


-- SIP/202-0000000d answered SIP/100-0000000c

Audio is at 38862
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 123.237.132.112:5010 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.22:5010;branch=z9hG4bK-d8754z-8357cb36704ef865-1—d8754z-;received=123.237.132.112;rport=5010
From: "100"sip:100@192.168.1.197:5060;tag=9e3f3209
To: sip:202@192.168.1.197:5060;tag=as499813a9
Call-ID: Y2FjMjkyMGQzNTdjYTkxN2EyMmEzM2JjNmEzMGVkOWU.
CSeq: 2 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@117.253.175.12:5060
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 353334487 353334487 IN IP4 117.253.175.12
s=Asterisk PBX 11.5.1
c=IN IP4 117.253.175.12
t=0 0
m=audio 38862 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34

<------------>
Retransmitting #1 (no NAT) to 123.237.132.112:5010:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.22:5010;branch=z9hG4bK-d8754z-8357cb36704ef865-1—d8754z-;received=123.237.132.112;rport=5010
From: "100"sip:100@192.168.1.197:5060;tag=9e3f3209
To: sip:202@192.168.1.197:5060;tag=as499813a9
Call-ID: Y2FjMjkyMGQzNTdjYTkxN2EyMmEzM2JjNmEzMGVkOWU.
CSeq: 2 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@117.253.175.12:5060
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 353334487 353334487 IN IP4 117.253.175.12
s=Asterisk PBX 11.5.1
c=IN IP4 117.253.175.12
t=0 0
m=audio 38862 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34


Retransmitting #2 (no NAT) to 123.237.132.112:5010:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.22:5010;branch=z9hG4bK-d8754z-8357cb36704ef865-1—d8754z-;received=123.237.132.112;rport=5010
From: "100"sip:100@192.168.1.197:5060;tag=9e3f3209
To: sip:202@192.168.1.197:5060;tag=as499813a9
Call-ID: Y2FjMjkyMGQzNTdjYTkxN2EyMmEzM2JjNmEzMGVkOWU.
CSeq: 2 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@117.253.175.12:5060
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 353334487 353334487 IN IP4 117.253.175.12
s=Asterisk PBX 11.5.1
c=IN IP4 117.253.175.12
t=0 0
m=audio 38862 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34


Retransmitting #3 (no NAT) to 123.237.132.112:5010:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.22:5010;branch=z9hG4bK-d8754z-8357cb36704ef865-1—d8754z-;received=123.237.132.112;rport=5010
From: "100"sip:100@192.168.1.197:5060;tag=9e3f3209
To: sip:202@192.168.1.197:5060;tag=as499813a9
Call-ID: Y2FjMjkyMGQzNTdjYTkxN2EyMmEzM2JjNmEzMGVkOWU.
CSeq: 2 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@117.253.175.12:5060
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 353334487 353334487 IN IP4 117.253.175.12
s=Asterisk PBX 11.5.1
c=IN IP4 117.253.175.12
t=0 0
m=audio 38862 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34


Retransmitting #4 (no NAT) to 123.237.132.112:5010:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.22:5010;branch=z9hG4bK-d8754z-8357cb36704ef865-1—d8754z-;received=123.237.132.112;rport=5010
From: "100"sip:100@192.168.1.197:5060;tag=9e3f3209
To: sip:202@192.168.1.197:5060;tag=as499813a9
Call-ID: Y2FjMjkyMGQzNTdjYTkxN2EyMmEzM2JjNmEzMGVkOWU.
CSeq: 2 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@117.253.175.12:5060
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 353334487 353334487 IN IP4 117.253.175.12
s=Asterisk PBX 11.5.1
c=IN IP4 117.253.175.12
t=0 0
m=audio 38862 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34


Retransmitting #5 (no NAT) to 123.237.132.112:5010:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.22:5010;branch=z9hG4bK-d8754z-8357cb36704ef865-1—d8754z-;received=123.237.132.112;rport=5010
From: "100"sip:100@192.168.1.197:5060;tag=9e3f3209
To: sip:202@192.168.1.197:5060;tag=as499813a9
Call-ID: Y2FjMjkyMGQzNTdjYTkxN2EyMmEzM2JjNmEzMGVkOWU.
CSeq: 2 INVITE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:202@117.253.175.12:5060
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 353334487 353334487 IN IP4 117.253.175.12
s=Asterisk PBX 11.5.1
c=IN IP4 117.253.175.12
t=0 0
m=audio 38862 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34

You are responding from a private address, which means Asterisk has not been told how to find your public address.