It seems that you SIP provider detects your NAT settings and it is
trying to fix it, which breaks your setup.
See the Contact in the 200OK: Contact:
<sip:0390156...@kamailio.int:5070;transport=udp>
Which is bad, it should be the IP of the FS server.
See the RURI in ACK: sip:0390156...@kamailio.ext:5061;transport=udp
The IP was switched from int to ext.
Probably a better setup would be to listen on multiple interfaces, one
for external traffic and one for internal traffic.
And then you will need to ask your SIP provider to stop messing with
the Contact header or you will need to circumvent that by doing some
header and routing manipulations.
Regards,
Ovidiu Sas
On Thu, Oct 16, 2014 at 9:09 PM, Nicholas Gill <n...@etellicom.com> wrote:
> Hello sr-users,
>
> We have Kamailio (behind NAT) configured acting as a proxy in front of some
> FreeSWITCH servers.
>
> There appears to be something amiss with my inbound (dispatcher)
> configuration which leads to misaddressing / misdelivery of the ACK after
> 200 OK (outbound calls appear to be proxied correctly [1]).
>
> Calls from the sip provider incoming to Kamailio are distributed by the
> dispatcher module to the FS server (For testing in this simplified scenario
> FS and Kamailio are on the same machine):
>
> SIP provider (5060) sends an invite to Kamailio(5061), Kamailio uses the
> dispatcher module to select a backend server and forwards the INVITE to
> FS(5070) (see inbound-callflow.png; kamailio.ext and kamailio.int are the
> same machine, just public/private addresses).
>
> I notice that at no point is a Via/record-route header for the FS server
> inserted into the forwarded session. I'm not actually certain this is a
> requirement, but I can't think of another obvious way that Kamailio could
> proxy the same session to the same FS server.
>
> The 100 Trying seems unremarkable [2], I suspect the 200 OK [3] is
> problematic. It has been proxied from the FS server, however contains no
> reference to the FS server address either in the via headers nor
> record-route (kamailio.int:5070). The incoming ACK [4] then appears to be
> misdelivered / lost - Kamailio receives it on the private address and
> forwards it to the public address rather than the FS server.
>
> My configuration [5] is built based on the default configuration + examples
> from the dispatcher module. There are some provisions for FreeSWITCH
> internal/external profiles made so the configuration listens on 2 different
> ports. This particular scenario should only use the 5061 port as it involves
> calls to an external sip provider (briefly 5060 should be proxied to FS:5080
> and 5061 should be proxied to FS:5070 and vice-versa).
>
>
> If someone can see an issue with the configuration and/or point to an error
> in the call flow (i.e. should FS be inserting the Via header?) that would be
> greatly appreciated.
>
>
> Thanks,
>
> -nick
>
> [1] Outbound call flow (see also outbound-callflow.png)
> FS(port 5070) sends an invite to Kamailio(5061) (Kamailio is configured in
> FS as an outbound proxy), INVITE contains Via header for the FS server,
> Kamailio forwards to sip provider, and routes all messages back and forth
> correctly.
>
> [2] 100 Trying
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP
> sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.0;rport=5060
> Via: SIP/2.0/UDP
> far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK
> From: "Nicholas Gill" <sip:0384171...@far.external.ip>;tag=j3KQmpvmg6mvr
> To: <sip:0390156...@sip.provider.com>
> Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9
> CSeq: 66413636 INVITE
> Server: kamailio (4.1.6 (x86_64/linux))
> Content-Length: 0
>
> [3] Proxied 200 OK (Kamailio -> Sip Provider)
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> sip.provider.com:5060;rport=5060;branch=z9hG4bKfffb.7c81ee53.0
> Via: SIP/2.0/UDP
> far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK
> Record-Route: <sip:kamailio.ext:5061;lr=on>
> Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr;did=0ec.84ff82b1>
> From: "Nicholas Gill" <sip:0384171...@far.external.ip>;tag=j3KQmpvmg6mvr
> To: <sip:0390156...@sip.provider.com>;tag=BFvQmggHrg74m
> Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9
> CSeq: 66413636 INVITE
> Contact: <sip:0390156...@kamailio.int:5070;transport=udp>
> User-Agent: IMX
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 270
> X-FS-Support: update_display,send_info
>
> [4] Incoming ACK (Sip Provider -> Kamailio)
> ACK sip:0390156...@kamailio.ext:5061;transport=udp SIP/2.0
> Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr>
> Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.2
> Via: SIP/2.0/UDP
> far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK509ZQBKB947aF
> Route: <sip:kamailio.ext:5061;lr=on>
> Max-Forwards: 69
> From: "Nicholas Gill" <sip:0384171...@far.external.ip>;tag=j3KQmpvmg6mvr
> To: <sip:0390156...@sip.provider.com>;tag=BFvQmggHrg74m
> Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9
> CSeq: 66413636 ACK
> Contact: <sip:gw+exe...@far.external.ip:5060;transport=udp;gw=Exetel>
> Proxy-Authorization: Digest username="0384171900", realm="far.external.ip",
> nonce="xxxx", algorithm=MD5, uri="sip:0390156...@sip.provider.com",
> response="xxxx"
> Content-Length: 0
> P-hint: rr-enforced
>
> [5] Kamailio configuration (based on default configuration). Dispatcher set
> 1 is FS internal profile (5080) and set 2 is external profile (5070)
> ...
> #!define WITH_NAT
> ...
> alias="kamailio.int:5060"
> alias="kamailio.int:5061"
> alias="kamailio.ext:5060"
> alias="kamailio.ext:5061"
>
> listen=udp:kamailio.int:5060 advertise kamailio.ext:5060
> listen=udp:kamailio.int:5061 advertise kamailio.ext:5061
> port=5060
> port=5061
> ...
> loadmodule "dispatcher.so"
> modparam("dispatcher", "flags", 2)
> modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list")
> modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")
> modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")
> modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")
> #modparam("dispatcher", "ds_ping_interval", 60)
> ...
> modparam("nathelper", "sipping_from", "sip:pin...@etellicom.com")
> ...
> request_route {
> ...
> # Not from FS servers gets dispatched to FS servers.
> if(!ds_is_from_list("1") and !ds_is_from_list("2"))
> {
> route(DISPATCH);
> }
>
> # Requests from FS External profile routed out via specific port.
> if(ds_is_from_list("2"))
> $fs="udp:kamailio.int:5061";
> ...
> }
>
> route[DISPATCH] {
> if(dst_port==5061)
> {
> $fs="udp:kamailio.int:5061";
> # Hash over to uri for test purposes
> if(!ds_select_dst("2", "2"))
> {
> send_reply("404", "No destination");
> exit;
> }
> }
> else
> {
> # Hash over to uri for test purposes
> if(!ds_select_dst("1", "2"))
> {
> send_reply("404", "No destination");
> exit;
> }
> }
>
> xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
> t_on_failure("RTF_DISPATCH");
> route(RELAY);
> exit;
> }
> failure_route[RTF_DISPATCH] {
> if (t_is_canceled()) {
> exit;
> }
> # next DST - only for 500 or local timeout
> if (t_check_status("500")
> or (t_branch_timeout() and !t_branch_replied()))
> {
> if(ds_next_dst())
> {
> t_on_failure("RTF_DISPATCH");
> route(RELAY);
> exit;
> }
> }
> }
> ...
>
>
> _______________________________________________
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>
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VoIP Embedded, Inc.
http://www.voipembedded.com
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