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jcolp at digium.com
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 Posted: Wed Mar 02, 2016 8:00 pm Post subject: [asterisk-users] RTP / NAT question ( pjsip ) |
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Kevin Long wrote:
| Quote: |
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.
I do have rewrite_contact=yes; on in my pjsip endpoint
configuration, but still the “rtp set debug on” command is showing
me that when I dial into the echo application, RTP packets are being
sent to the private IP and not the public IP .
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The "rtp_symmetric" option is used to control this for RTP. When set to
yes media will be sent to the source IP address+port of the received RTP.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
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jcolp at digium.com
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| Posted: Wed Mar 02, 2016 8:17 pm Post subject: [asterisk-users] RTP / NAT question ( pjsip ) |
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Kevin Long wrote:
| Quote: | 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 on asterisk server showing UDP packet bound for my remote
endpoints internal IP: 17:07:57.130212 IP 10.50.55.10.6214>
10.128.30.239.51126: UDP, length 182
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It needs to be set to yes, and you also must have received an RTP packet
from the remote side. If you haven't received any then check to see if
they are being blocked by a firewall, and that the SDP sent to the
device contains the public IP address.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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jcolp at digium.com
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| Posted: Thu Mar 03, 2016 6:52 am Post subject: [asterisk-users] RTP / NAT question ( pjsip ) |
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Kevin Long wrote:
| Quote: | 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 transport= into my
endpoint configuration hoping they will “inherit” the correct public
IP for the media .
But Asterisk is still sending RTP to the wrong IP .
I am trying to test a “real world” scenario of public IP and NAT
traversal, but I do have split tunnel VPN in my environment so the
endpoint and the asterisk server *could* reach each other by the
private IP ,but I am actually trying to avoid this with a proper
configuration since my real users will not be on any VPN, mostly.
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What version of 13 are you also using?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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jcolp at digium.com
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