21 Jan
2014
21 Jan
'14
2:24 p.m.
Hi Fred,
I have followed your HOWTO and the scenario remains exactly the same.
I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP to Asterisk
IP, and back directly to Phone2 public IP.
I might be making wrong assumptions regarding this traffic flow. Is that correct?
Thank you
21 Jan
21 Jan
3:06 p.m.
New subject: Kamailio behind NAT
Hi John,
rtpproxy is not enough if you are using asterisk in your environment.
You have to check that asterisk is configured to work with NAT, otherwise you will
experience audio problems.
Are the asterisk RTP ports enabled/forwarded on your firewall?
Regards,
Kostas
On Jan 21, 2014, at 2:24 PM, John Smith <jsmith.15(a)mail.com> wrote:
Hi Fred,
I have followed your HOWTO and the scenario remains exactly the same.
I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP to Asterisk
IP, and back directly to Phone2 public IP.
I might be making wrong assumptions regarding this traffic flow. Is that correct?
Thank you
_______________________________________________
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
3:28 p.m.
New subject: Kamailio behind NAT
Actually, it should work without any NAT traversal done in Asterisk, if
Asterisk communicates never direct with the phones, but only via
Kamailio and rtpproxy. In this case, Asterisk can use private IP
addresses. All the near-end NAT traversal can be done in Kamailio.
regards
Klaus
On 21.01.2014 14:06, meres wrote:
Hi John,
rtpproxy is not enough if you are using asterisk in your environment.
You have to check that asterisk is configured to work with NAT, otherwise you will
experience audio problems.
Are the asterisk RTP ports enabled/forwarded on your firewall?
Regards,
Kostas
On Jan 21, 2014, at 2:24 PM, John Smith <jsmith.15(a)mail.com> wrote:
Hi Fred,
I have followed your HOWTO and the scenario remains exactly the same.
I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP to Asterisk
IP, and back directly to Phone2 public IP.
I might be making wrong assumptions regarding this traffic flow. Is that correct?
Thank you
_______________________________________________
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
_______________________________________________
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
3:25 p.m.
New subject: Kamailio behind NAT
On 21.01.2014 13:24, John Smith wrote:
I might be making wrong assumptions regarding this
traffic flow. Is that correct?
That depends on your policy. It is up to you to define how RTP should be
routed. There are basically 2 choices:
a) RTP from clients is handled by rtpproxy:
phone1 <-nat-> rtpproxy <--> Asterisk <--> rtpproxy <-nat->
phone2
In this case, only the private IP of Kamailio and rtpproxy (can be the
same IP address) must be mapped to a public IP address.
b) RTP directly to Asterisk:
phone1 <-nat-> Asterisk <-nat-> phone2
In this case, the private IPs of Kamailio and Asterisk must be mapped to
a public IP address.
When using version a) you have to make sure to set the proper IP address
in the SDP. For example, SDPs in messages from Kamailio to the phone
must contains the PUBLIC IP of rtpproxy in the c=... line. SDPs in
messages from Kamailio to Asterisk must contain the PRIVATE IP of
rtpproxy in the c=... line.
regards
Klaus
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days inactive
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3 comments
3 participants
participants (3)
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John Smith -
Klaus Darilion -
meres