does the SIP request gets to client? Can you paste here the INVITE?Hi,
I started working on IPv4/IPv6 Translation issue in Kamailio by using rtpproxy as Media relay last year in my Bachelor-project.
I solved the trouble with Translation between IPv4 and IPv6 clients, where IPv4 client used public IPv4 address. Now I am working on the issue, where
IPv4 client uses private IPv4 (is behind NAT in local network), so I have to provide IPv4/v6 and NAT for all messages in one call.
My topology consists of:
Kamailio 3.3.4 - SIPproxy (supports IPv4/IPv6, public IPv4 and IPv6 addresses) - makes NAT translation, works as SIPoutbound proxy and communicates with rtpproxy (v1.2.1),
Kamailio 3.3.4 - SIPprotocol GW (== SIP bridge) - translates IPv4 messages to IPv6 (and IPv6 to IPv4), uses other rtpproxy, has public IPv4 and IPv6 addresses
+ only IPv4 and only IPv6 clients (Linphone v3.5.2)
I assume, that both (IPv4 and IPv6) clients are in the same SIP domain.
I solve IPv4-IPv6 translation so, that each message from IPv6 client goes to SIP proxy and (if needed) is forwarded to SIP Protocol GW. Translated medssage is then forwarded back to SIP proxy and from that to callee.
Here i have some problems:
-1.) IPv6 client doesnt respond to invites from IPv4 client. The messages are ignored at IPv6 client.
This sounds like mis-routed ACK. RTPPRoxy does not clear sessions by itself.
-2.) Rtpproxy on the machine with SIP protocol gateway (which provides IPv4/v6 translation) removes session earlier than I need (usually after 30 seconds of call) and the result is call break.
this issue doesnt appear for rtpproxy on SIP proxy. I dont know what causes this behavior, if it is only in some parameters, witch i have to set, or it is bug in rtpproxy or ....
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda