Re: [SR-Users] audio issue in same nat.

oh now getting worst when i test a few more times both 101 call 102 vice versa both no audio.but 102 got audio when call voicemail.101 still no audio when call voicemail. ReGaRds, MinGh0n From: gminghon(a)hotmail.com To: sr-users(a)lists.sip-router.org Subject: audio issue in same nat. Date: Wed, 22 Jun 2011 17:34:53 +0800 Hi List, below is my setup.. rtpproxy and kamailio in one PC with 2 nic. (ppp0 with public IP[60.49.119.XX] and eth1 with private IP[192.168.2.3])and asterisk is on another PC with private IP[192.168.2.23] i use realtime integration for kamailio and asterisk.http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6… i have two yealink hardphone ext 101(ip 192.168.1.200) and 102(ip 192.168.1.132) and a softphone ext 103 registered successful.both hardphone are behind same nat (175.136.221.XX)and softphone ext 103(ip 10.129.138.225) behind nat also (113.210.97.XX) ul show:database engine 'MYSQL' loadedControl engine 'FIFO' loadedentering fifo_cmd ul_dumpDomain:: location table=512 records=3 max_slot=1 AOR:: 102 Contact:: sip:102@175.136.221.XX:5062 Q= Expires:: 3109 Callid:: 2043273564(a)175.136.221.241 Cseq:: 4 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.241:1039 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.49.119.69:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@113.210.97.XX:58776;transport=UDP;ob Q= Expires:: 294 Callid:: oa8Pqx3mR.SVnzAVEYHTwVKZE8CbpY9l Cseq:: 27626 User-agent:: v1.0.0/iPhone State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 8143 AOR:: 101 Contact:: sip:101@175.136.221.XX:5062 Q= Expires:: 1738 Callid:: 451417581(a)175.136.221.241 Cseq:: 2 User-agent:: T20 9.41.0.80 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 16383FIFO command was::ul_dump:openser_receiver_17783 103 try to call 102 and 101 work fine. 101 and 102 try call 103 also fine.when 101 call 102 it work fine but when 102 call 101 there is no audio for both side.102 call 101 wireshark capture on 102 sidekeep send rtp but no receive.192.168.1.132 -> 60.49.119.XX RTP when capture on 101 side.keep send rtp but no receive.192.168.1.200 -> 60.49.119.XX RTP and also when 101 try to call into voicemail there is no audioit keep send rtp packet but to192.168.1.200 -> 192.168.2.23 RTP in kamailio.cfg#!WITH_NATlisten=60.49.119.XXlisten=192.168.2.3 # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); nat_uac_test("19")rtpproxy -l 60.49.119.XX -s udp:127.0.0.1 is running anyone can help me? how can i fix this?thanks in adv. Regards, minghon