I'm already following the documentation. But as I answer the question from Tony, right now I used static NAT and open all the port. I will give it a try to do the port forwarding
Thanks Robert I will update the result > -----Original Message----- > From: Robert Joly [mailto:[email protected]] > Sent: Friday, May 15, 2009 9:17 PM > To: Boy Aidil Sjam; [email protected] > Subject: RE: [sipx-users] Media Relay Service Failed > > > Right now, I'm still open all UDP & TCP port for the SIPX server. > > I used static ip address for the SIPX server NAT > > > > > > > -----Original Message----- > > > From: Robert Joly [mailto:[email protected]] > > > Sent: Friday, May 15, 2009 8:25 PM > > > To: Boy Aidil Sjam; [email protected] > > > Subject: RE: [sipx-users] Media Relay Service Failed > > > > > > > Hi All, > > > > I just found out that, after I put back the NAT port to default > > > > configuration (30000-31000), the remote worker support can't run > > > > anymore. > > > > Internal user able to called remote user and vice versa. When the > > > > call established remote user can hear the voice from > > internal user > > > > but internal user can't hear anything. > > > > > > > > Here the packet stream that I've got from the trace file (capture > > > > from both side WAN & LAN) > > > > > > > > Remote user RTP packet stream > > > > Remote Worker (public address) <----- RTP ---> SIPX > > (public address) > > > > > > > > Internal user RTP packet stream > > > > Internal user (private address) <---- RTP ---> SIPX (private > > > > address) > > > > > > > > Both using the same codec G.711 > > > > > > > > If I change back the NAT port to 10000-20000 (which is > > > > overlap/conflict with java application port 15060) and the Media > > > > service status failed (obviously), after the called > > established both > > > > end can hear clearly. > > > > > > > > But I saw the different RTP packet stream with this configuration > > > > > > > > Remote Worker (public address) <---- RTP ---> Internal > > user (private > > > > address) > > > > > > > > Can someone help me with this problem? > > > > > > > > B.Aidil > > > > > > When you change the RTP port range used by the Media Relay you also > > > need to change the firewall pinholes and port forwarding > > rules on your > > > NAT/Firewall to line up with that new port range. Have you > > done that? > > > So, just to make sure I'm getting this straight, if you set the range > to > 10000-20000, sipXrelay fails to start but you remote worker calls work > and when you set the range to 30000-31000, sipXrelay comes up fine but > the remote worker call gives you one-way speech path... This result is > completely surpizing and certainly goes against logic. siPXrelay is > supposed ot help, not hinder. Do me a favor, please restore your media > port range to 30000-31000 and go to > http://sipx- > wiki.calivia.com/index.php/Configuring_remote_workers_cheats > heet and make sure you follow each and every rule to the line. > Spceifically, pay attention to step #4 which you do not quite follow > yet > and also make sure that you disable each and every NAT traversal > assistance features that may be present in components outside of sipX. > Make sure that any STUN and/or TURN and/or ICE and/or SIP ALGs is > disabled in your remote phones, remote firewall and local firewall. If > the problem remains after this, please send me the LAN & WAN traces > off-list. > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.5.329 / Virus Database: 270.12.30/2115 - Release Date: > 05/14/09 17:54:00 DISCLAIMER: Information in this message is confidential and may be legally privileged. It is intended solely for the addressees. Access to this message by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
