[SR-Users] audio issue in same nat.

[†ஐû●чƃцίш† Goh]

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-1.6.2-astdb
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 at 175.136.221.XX:5062 Q=                        Expires:: 3109                        Callid:: 2043273564 at 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 at 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 at 175.136.221.XX:5062 Q=                        Expires:: 1738                        Callid:: 451417581 at 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 		 	   		  
 		 	   		  
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