Hi Vivek, Thanks for the link. I had a look through and couldn't find anything that worked. There are no NAT problems as this is all taking place on my internal network. The rtp.conf is used to configure the ports. There are no firewalls or gateways in between these devices.
Asterisk is listening on the correct ports, and receiving the traffic, as no ICMP messages are being generated to say that the packets could not be delivered. Ryan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vivek Shrivastava Sent: Monday, 12 November 2007 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded well i think rtp port range is defined in rtp.conf and correct me if i am wrong, these ports must be opened/forwarded to communicate. http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Let me know if you need more information. Thanks, Vivek On 11/11/07, Ryan Newington <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> wrote: Hi Vivek, I'm not sure what you mean, could you explain further? Regards Ryan From: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>] On Behalf Of Vivek Shrivastava Sent: Monday, 12 November 2007 1:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, I was just wondering if they need to be according rtp.conf. ( or you may need to modify rtp.conf) Regards, Vivek On 11/11/07, Ryan Newington <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> wrote: Hi Vivek, The SIP port is set to the standard port 5060. The RTP ports as far as I know are random ephemeral ports between 63000 and 64000. I can change the port range on the media server, asterisk and the device, but neither seems to help. My diagram below is probably misleading. The RTP traffic flow that I see is as follows (one way traffic into Asterisk) SIP Phone <---> Media Gateway ---> Asterisk <--- SIP Phone Ryan From: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>] On Behalf Of Vivek Shrivastava Sent: Sunday, 11 November 2007 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking random port selection option on the device/softphone may help. --Vivek On 11/10/07, Ryan Newington <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> wrote: Hi Luki, Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success. Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway. SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone An asterisk internal call will work fine. Eg; SIP Phone <-> Asterisk <-> SIP Phone Regards Ryan -----Original Message----- From: [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> [mailto: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ] On Behalf Of Luki Sent: Sunday, 11 November 2007 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded > When using 'rtp debug' on the asterisk console, it shows that it is > receiving traffic from one endpoint, but not the other. A wireshark trace > reveals it is actually receiving traffic from both ends. Sounds like a firewall issue. Wireshark shows what's "on the wire", i.e. before iptables. The packets are being dropped for whatever reason and never reach the asterisk process. Check your iptables and RTP port range, and perhaps try changing it. Luki _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com--<http://www.api-digital.com--/> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- <http://www.api-digital.com--/> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- <http://www.api-digital.com--/> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com--<http://www.api-digital.com--/> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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