As it turns out, we did, in fact set the flags wrong: We didn't set them at all!

In route[NATMANAGE] we were missing a crucial 'b' in isBflagset(FLBV4V6), leading to the RTPProxy flags never being set to FAIE or FAEI.

A diff (shame on us for not trying it first) revealed the mistake, and now all is well.
On 12/03/2013 05:17 PM, Mark Zeman wrote:
The SIP packets are fine - we are able to establish SIP session with a 
secured control channel as easily as with an unsecured one.
As long as the RTPProxy doesn't get involved (i.e. as long as it's 
IPvX-IPvX and not IPv4-IPv6) everything works!
With the RTPProxy, the connection gets established (SIP still works) 
but RTP only comes from one side, the caller, and even that is not 
relayed.

It appears that Kamailio doesn't even enter the route[IPV4V6] part of 
its config, for example...

I will be able to send logs tomorrow, but I can show you our config, if 
that might help.





On Tue 03 Dec 2013 03:27:55 PM CET, Klaus Darilion wrote:

On 03.12.2013 14:23, Mark Zeman wrote:

Hello all,

The subject says most of it, I think.

We set up our Kamailio and RTPProxy according to
http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
with the addition of an alias (siplab.ch), and the DNS to go with it, as
well as TLS and SRTP.

However, we only get working calls IPv4-IPv4 and IPv6-IPv6! IPv4-IPv6 we
get a proper connection,
secured with SRTP, but no audio. Looking at the network, RTP packets go
from the caller to the server,
but nothing leaves the server and no RTP packets go from callee to
server.

Do you have any idea how to fix this?

Before fixing you need to find the problem. Probably the SDP get
rewritten incorrect. To debug this issue, you have to inspect the ip
address and ports in the received and sent SDPs if they are correct
(e.g. IPv4 and IPv6 address of the rtpproxy).

You mentioned TLS - thus it is difficult to inspect the raw SIP
packets. Is such cases you can either:
- adding lots of xlog("$mb, ...") to your config and watch syslog
- use the sipcapture module to write every incoming and outgoing SIP
message to the DB, and analyze the packets in the DB
- Disable TLS. Use TCP. If it fails with TCP it will also fail with
TLS. Once it works over TCP, it will also work over TLS.
- If you really have a problem with TLS only, you could enable the
NULL cipher to have a unencrypted TLS connection, or use Wireshark's
TLS decoding capabilities
- Configure rtpproxy to communicate via UDP socket instead TCP socket.
Then you can also capture the communication between Kamailio and
rtpproxy.

It seems you just call the rtpproxy functions with some wrong flags.
Make sure properly detect the direction (4to6 or 6to4) an set the i
and e flags accordingly.

regards
Klaus



_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users