Shnuer, The issue is that when the packet comes from behind the Baudtech Asterisk see's an external IP and there for thinks there is no NAT and is looking for the RTP on the ports stated. When you send from the Linksys it see's a local IP and knows there is NAT. Do you have nat=yes in sip.conf in both the general section and for the IP of OpenSIpS?
Rergards, dOvid -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Schneur Rosenberg Sent: Monday, January 30, 2012 15:07 To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] No audio on some routers with PAP2T Hi, I have a openSIPS server setup to do registration and load balancing between 2 Asterisk servers, the Asterisk servers do everything besides registration and they are load balanced by the openSIPS servers, incoming calls hit the openSIPS server which sends it to the Asterisk server and if it needs to go to a local phone it sends it back to openSIPS where the phone is registered to, outgoing calls get sent to Asterisk via load balancing and asterisk completes the call. I have a problem with some ata's (in my case pap2t) that when its behind certain routers (in my case a Baudtech) there is no audio, when I try a different router it does work, also when I try a different ata like a spa2102 it does work, also when I connect the pap2t directly to the asterisk it works fine, NONE of the routers have SIP ALG enabled, it seems that the nat blocks the audio when the media is from a different server. The interesting thing is that the Baudtech router changes the internal IP's to the external ip, the other router does not, does that mean that there is some kind of ALG built into the Baudtech router? even if it does how come the Asterisk server handles the audio fine while the openSIPS breaks the audio here is a trace of the initial INVITE from the Baudtech (the problematic one) as you can see the ip at the Via and Contact and in the c tag in the RTP have been replaced by the router U 46.116.60.131:5060 -> 64.69.33.43:5060 INVITE sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 46.116.60.131:5060;branch=z9hG4bK-278a1ace;rport. From: <sip:[email protected]>;tag=aea4a93b350d6746o0. To: <sip:[email protected]>. Call-ID: [email protected]. CSeq: 101 INVITE. Max-Forwards: 70. Contact: <sip:[email protected]:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 442. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 50374 50374 IN IP4 46.116.60.131. s=-. c=IN IP4 46.116.60.131. t=0 0. m=audio 16476 RTP/AVP 0 2 4 8 18 96 97 98 100 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. Here is the same invite when send from the DLINK router (here audio is fine) U 85.250.89.78:5060 -> 64.69.33.43:5060 INVITE sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK-65cff5ef;rport. From: <sip:[email protected]>;tag=2ab40bee91703297o0. To: <sip:[email protected]>. Call-ID: [email protected]. CSeq: 101 INVITE. Max-Forwards: 70. Contact: <sip:[email protected]:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 442. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 19246 19246 IN IP4 192.168.2.100. s=-. c=IN IP4 192.168.2.100. t=0 0. m=audio 16438 RTP/AVP 0 2 4 8 18 96 97 98 100 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
