Re: [asterisk-users] RTP / NAT question ( pjsip )

2016-03-03 Thread Kevin Long
So the patch did resolve the audio RTP issue and I can make echo calls now, but it seems like the last issue I posted to the list, (pjsip driver making new outbound TLS transports instead of using existing SIP connection, not NAT friendly) is happening again .. Could that be? Thanks

Re: [asterisk-users] RTP / NAT question ( pjsip )

2016-03-03 Thread Kevin Long
Hi Joshua, This Asterisk 13 was pulled from git master branch just 2-3 days ago: GIT-13-d1495b . I used this very recent source code to overcome a pjsip problem (you can see my email list post from a few days ago) Thanks again smime.p7s Description: S/MIME cryptographic signature --

Re: [asterisk-users] RTP / NAT question ( pjsip )

2016-03-03 Thread Joshua Colp
Kevin Long wrote: Hi Joshua, Looking at the transmitted SIP packets from Asterisk, it looks like Asterisk is only sending it’s own internal IP (it is behind a NAT too, with proper port forwarding) . I did set in my transport the external_signaling_address and external_media_address , and

Re: [asterisk-users] RTP / NAT question ( pjsip )

2016-03-02 Thread Kevin Long
Hi Joshua, Looking at the transmitted SIP packets from Asterisk, it looks like Asterisk is only sending it’s own internal IP (it is behind a NAT too, with proper port forwarding) . I did set in my transport the external_signaling_address and external_media_address , and I have now put

Re: [asterisk-users] RTP / NAT question ( pjsip )

2016-03-02 Thread Joshua Colp
Kevin Long wrote: Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the

Re: [asterisk-users] RTP / NAT question ( pjsip )

2016-03-02 Thread Kevin Long
Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump

Re: [asterisk-users] RTP / NAT question ( pjsip )

2016-03-02 Thread Joshua Colp
Kevin Long wrote: I am having trouble with RTP and NAT : Below is a SIP SDP invite from a remote endpoint which is trying to call extension 420 which is the ECHO application . As you can see, the public IP is where the request comes in from, but the SDP contains the private, internal IP in

[asterisk-users] RTP / NAT question ( pjsip )

2016-03-02 Thread Kevin Long
I am having trouble with RTP and NAT : Below is a SIP SDP invite from a remote endpoint which is trying to call extension 420 which is the ECHO application . As you can see, the public IP is where the request comes in from, but the SDP contains the private, internal IP in numerous places.