Re: [asterisk-users] Dropping RTP packets
You need canreinvite=no in the config for your sip phone and the veracity connection, otherwise Asterisk will just mediate the call setup then try to allow the sip phone and veracity to talk directly to one another. Jim Dickenson wrote: I have a SIP phone at home behind a NAT router registered with an * box at my office with a routable static IP address running version SVN-branch-1.6.0-r175638M. If I make a call from my SIP phone out a PRI circuit to my cell phone everything works as expected. I hear audio in both directions and all is good. If from the same SIP phone I make a call via our Veracity SIP account to my cell phone I hear no audio in either direction. In trying to find out what is wrong I used tcpdump to see if I could learn anything. I can see the phone sending fixed length UDP packets on to my home network heading to the IP address of the * box. If I run tcpdump on the * box I do not see the packets being received. I do not see the * box sending any packets to my home network either. I have not checked if the * box is receiving packets from Veracity I only know that no audio packets are sent to my home network. If I use tcpdump to watch the SIP phone call via the PRI circuit I see packets both on my home network and my * box. If I use a SIP phone located in my office and make a call via Veracity everything is okay. Also a co-worker has a vpn router on his home network connected to the office vpn server and he can make calls from his SIP phone via Veracity without problems. I can also call his SIP phone from my SIP phone and packets pass as expected. It seems as if audio packets from my SIP phone disappear only if they are involved with a call via Veracity. Does anyone have some idea what I might look at to find what is causing this problem? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping RTP packets
The problem turned out to be a firewall issue. If one makes a call out the PRI line * sends the ring audio to the sip phone. This opened a hole in the firewall for the return traffic so things worked. If I make the call from my office when the call was answered and the caller started talking and that opened the hole. As no traffic went out the firewall no dynamic hole was created so nothing passed. I had remembered a previous router's configuration that allowed any UDP traffic and my current one is setup different. One problem with working at this for 33 years and remembering things from the past that are different now. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Brent Davidson br...@texascountrytitle.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 26 Feb 2009 15:18:14 -0600 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dropping RTP packets You need canreinvite=no in the config for your sip phone and the veracity connection, otherwise Asterisk will just mediate the call setup then try to allow the sip phone and veracity to talk directly to one another. Jim Dickenson wrote: I have a SIP phone at home behind a NAT router registered with an * box at my office with a routable static IP address running version SVN-branch-1.6.0-r175638M. If I make a call from my SIP phone out a PRI circuit to my cell phone everything works as expected. I hear audio in both directions and all is good. If from the same SIP phone I make a call via our Veracity SIP account to my cell phone I hear no audio in either direction. In trying to find out what is wrong I used tcpdump to see if I could learn anything. I can see the phone sending fixed length UDP packets on to my home network heading to the IP address of the * box. If I run tcpdump on the * box I do not see the packets being received. I do not see the * box sending any packets to my home network either. I have not checked if the * box is receiving packets from Veracity I only know that no audio packets are sent to my home network. If I use tcpdump to watch the SIP phone call via the PRI circuit I see packets both on my home network and my * box. If I use a SIP phone located in my office and make a call via Veracity everything is okay. Also a co-worker has a vpn router on his home network connected to the office vpn server and he can make calls from his SIP phone via Veracity without problems. I can also call his SIP phone from my SIP phone and packets pass as expected. It seems as if audio packets from my SIP phone disappear only if they are involved with a call via Veracity. Does anyone have some idea what I might look at to find what is causing this problem? ___ -- Bandwidth and Colocation Provided by 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping RTP packets
I have a SIP phone at home behind a NAT router registered with an * box at my office with a routable static IP address running version SVN-branch-1.6.0-r175638M. If I make a call from my SIP phone out a PRI circuit to my cell phone everything works as expected. I hear audio in both directions and all is good. If from the same SIP phone I make a call via our Veracity SIP account to my cell phone I hear no audio in either direction. In trying to find out what is wrong I used tcpdump to see if I could learn anything. I can see the phone sending fixed length UDP packets on to my home network heading to the IP address of the * box. If I run tcpdump on the * box I do not see the packets being received. I do not see the * box sending any packets to my home network either. I have not checked if the * box is receiving packets from Veracity I only know that no audio packets are sent to my home network. If I use tcpdump to watch the SIP phone call via the PRI circuit I see packets both on my home network and my * box. If I use a SIP phone located in my office and make a call via Veracity everything is okay. Also a co-worker has a vpn router on his home network connected to the office vpn server and he can make calls from his SIP phone via Veracity without problems. I can also call his SIP phone from my SIP phone and packets pass as expected. It seems as if audio packets from my SIP phone disappear only if they are involved with a call via Veracity. Does anyone have some idea what I might look at to find what is causing this problem? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
