[Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
Hello everybody, I'm newto Asterisk and I'm trying to configure the SIP side. I use Asterisk under the following configuration: SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution? Regards, Alex.Help STOP spam with the new MSN 8 and get 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
This is done in the rtp.conf file. You specify the port range with a start and end number. By default the range is 1 through 2. Leif. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alexander Simeonidis Sent: Thursday, May 13, 2004 10:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages Hello everybody, I'm new to Asterisk and I'm trying to configure the SIP side. I use Asterisk under the following configuration: SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution? Regards, Alex. Help STOP spam with the new MSN 8 http://g.msn.com/8HMAEN/2731??PS=47575 and get 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages
Alex, The media ports are configured in rtp.conf. Also, note that Asterisk sends RTP packets out the same ports it expects them to return on. This has the effect of creating a NAT mapping for that 5-tuple, as well as opening a hole in your firewall (naturally, YMMV depending on exactly what you're running for a firewall). One interesting consequence of the way Asterisk works is that if you don't have anything behind the NAT/Firewall that's generating RTP packets (ie, no audio) no hole gets made and incoming packets will get rejected. I recently ran into an interesting problem with two SIP phones trying to talk through Asterisk behind a (non-NAT) firewall. The problem was both phones were sending RTP to the Asterisk box but the firewall was blocking both RTP streams because Asterisk never sent any RTP out those ports. And the reason Asterisk hadn't sent RTP out those ports was because it was waiting for RTP from each of the two SIP phones. This was the classic chicken-and-egg scenario. I resolved it by opening up the firewall for the range of ports I had configured Asterisk to use for RTP. A better solution would be fore Asterisk to always send a starter RTP packet so that it can ensure that the firewall opens up. -brian Alexander Simeonidis wrote: Hello everybody, I'm new to Asterisk and I'm trying to configure the SIP side. I use Asterisk under the following configuration: SIP Proxy INTERNET | NAT FIREWALL | Asterisk SIP Phone A I'm trying to put a call from SIP Phone A through Asterisk to the SIP Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed that the port used to deliver the audio changes randomly. I would like to fix that to a specific range of ports so that I can tell to NAT Firewall to port forward these particalar ports to Asterisk. I have searched on documentation and the only thing that I found was how to change the SIP port but not the media port. Has anybody any ideas on how to solve that problem or where to look for a solution? Regards, Alex. Help STOP spam with the new MSN 8 http://g.msn.com/8HMAEN/2731??PS=47575 and get 2 months FREE*___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
