[Users] Audio problem in NAT

Hello Users, I posted so many mail to users but no one reply my issue please help me openSER proxy is mysipdomain.com , and its private_ ip is 192.168.2.60 and SIP server and Proxy is also in Behind UAC's are Behind the NATs --------------------- INVITE --------------- U 59.144.88.7:5060 -> 192.168.2.60:5060 INVITE sip:9002@mysipdomain.com;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKe2a540a8170eb12a. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>. Call-ID: 1685867393(a)192.168.1.2. CSeq: 1 INVITE. Contact: Indian-2 <sip:8002@192.168.1.2:5060;user=phone;transport=udp>. User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A). Expires: 300. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: 100rel,replaces. Content-Length: 245. Content-Type: application/sdp. v=0. o=8002 14279 14279 IN IP4 192.168.1.2. s=ATA186 Call. c=IN IP4 192.168.1.2. t=0 0. m=audio 16386 RTP/AVP 0 4 8 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 61.17.248.68:3186 INVITE sip:9002@192.168.2.7:5060;user=phone;transport=udp SIP/2.0. Max-Forwards: 10. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: Indian-2 <sip:8002@59.144.88.7:5060;user=phone;transport=udp>. User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A). Expires: 300. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: 100rel,replaces. Content-Length: 265. Content-Type: application/sdp. v=0. o=8002 14329 14329 IN IP4 192.168.1.2. s=ATA186 Call. c=IN IP4 192.168.1.2. t=0 0. m=audio 16386 RTP/AVP 0 4 8 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=direction:active. ------------------- RINGING------------ U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Require: 100rel. RSeq: 1. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Content-Length: 0. . # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Require: 100rel. RSeq: 1. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Content-Length: 0. U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 61.17.248.68:3186 -> 192.168.2.60:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. . v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.2.60:5060 -> 59.144.88.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7 ;branch=z9hG4bKbd027751c869b9ff. Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>. From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537. To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912. Call-ID: 1685867393(a)192.168.1.2. CSeq: 2 INVITE. Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>. Server: Cisco ATA 188 v3.2.1 atasip (050616A). Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE. Supported: replaces. Content-Length: 193. Content-Type: application/sdp. v=0. o=9002 27865 27865 IN IP4 192.168.2.7. s=ATA186 Call. c=IN IP4 192.168.2.7. t=0 0. m=audio 16386 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com