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@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

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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