20 Dec
2013
20 Dec
'13
6:13 p.m.
Hey all,
**I apologize if this is a duplicate, I received a bounce back on my first
email.
I have kamailio set up behind a nat(port restricted, with firewall rules to
allow sip transactions and allowing rtpproxy packet handling if needed) on
Amazon EC2. I can register and calls complete, however, the Caller(the one
initiating the transaction) receives no rtp media feed. I am running with
NAT enabled on kamailio and have rtpproxy installed listening on the public
IP. Kamailio and the rtpproxy are communicating(I have verified via the
kamailio debug logs). If I make a call between the exact same voip machines
directly via local IP on the same Nat(skipping kamailio), the calls
complete and they both receive feeds.
Both the Caller(party making the call) and the Callee(party receiving the
call) are behind a Port Restricted Nat.
This is a folder containing the debug output for one of these calls and the
kamailio.cfg settings
https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp…
Quick FYI, the Caller Display Name and the Callee SIP UserName are the same
string. However, in my understanding about sip, the display name means
pretty much nothing and is just a human readable string that does not
effect packet transport. If I am wrong and should test with a different
display name, let me know.
Thank you for the assistance,
ben
20 Dec
20 Dec
6:26 p.m.
Hi Ben,
Some NATs require symmetric RTP, so that source and destination ports
are the same for incoming and outgoing media streams. This can be
configured using the 'w' flag to rtpproxy_manage in the NATDETECT route.
rtpproxy_manage("cow");
Regards,
Hugh
On 20/12/2013 16:13, Benjamin Trent wrote:
Hey all,
**I apologize if this is a duplicate, I received a bounce back on my
first email.
I have kamailio set up behind a nat(port restricted, with firewall
rules to allow sip transactions and allowing rtpproxy packet handling
if needed) on Amazon EC2. I can register and calls complete, however,
the Caller(the one initiating the transaction) receives no rtp media
feed. I am running with NAT enabled on kamailio and have rtpproxy
installed listening on the public IP. Kamailio and the rtpproxy are
communicating(I have verified via the kamailio debug logs). If I make
a call between the exact same voip machines directly via local IP on
the same Nat(skipping kamailio), the calls complete and they both
receive feeds.
Both the Caller(party making the call) and the Callee(party receiving
the call) are behind a Port Restricted Nat.
This is a folder containing the debug output for one of these calls
and the kamailio.cfg settings
https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp…
Quick FYI, the Caller Display Name and the Callee SIP UserName are the
same string. However, in my understanding about sip, the display name
means pretty much nothing and is just a human readable string that
does not effect packet transport. If I am wrong and should test with a
different display name, let me know.
Thank you for the assistance,
ben
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Hugh Waite
Principal Design Engineer
Crocodile RCS Ltd.
7:42 p.m.
Hugh,
Thank you for the response, however, now neither the callee or the caller
are getting a feed. I am not sure if the nat requires symmetric RTP. If
there is a way to determine that, please let me know.
Summery, even with Hugh's possible solution, this still does not work.
Thank you again,
ben
On Fri, Dec 20, 2013 at 10:26 AM, Hugh Waite
<hugh.waite(a)crocodile-rcs.com>wrote;wrote:
Hi Ben,
Some NATs require symmetric RTP, so that source and destination ports are
the same for incoming and outgoing media streams. This can be configured
using the 'w' flag to rtpproxy_manage in the NATDETECT route.
rtpproxy_manage("cow");
Regards,
Hugh
On 20/12/2013 16:13, Benjamin Trent wrote:
Hey all,
**I apologize if this is a duplicate, I received a bounce back on my
first email.
I have kamailio set up behind a nat(port restricted, with firewall rules
to allow sip transactions and allowing rtpproxy packet handling if needed)
on Amazon EC2. I can register and calls complete, however, the Caller(the
one initiating the transaction) receives no rtp media feed. I am running
with NAT enabled on kamailio and have rtpproxy installed listening on the
public IP. Kamailio and the rtpproxy are communicating(I have verified via
the kamailio debug logs). If I make a call between the exact same voip
machines directly via local IP on the same Nat(skipping kamailio), the
calls complete and they both receive feeds.
Both the Caller(party making the call) and the Callee(party receiving
the call) are behind a Port Restricted Nat.
This is a folder containing the debug output for one of these calls and
the kamailio.cfg settings
https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp…
Quick FYI, the Caller Display Name and the Callee SIP UserName are the
same string. However, in my understanding about sip, the display name means
pretty much nothing and is just a human readable string that does not
effect packet transport. If I am wrong and should test with a different
display name, let me know.
Thank you for the assistance,
ben
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Hugh Waite
Principal Design Engineer
Crocodile RCS Ltd.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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days inactive
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2 comments
2 participants
participants (2)
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Benjamin Trent -
Hugh Waite