[OpenSIPS-Users] OpenSIPS mhomed
Yuri Kirsanov
y.kirsanov at gmail.com
Tue Dec 7 14:17:11 CET 2010
Hi Bogdan,
Thank you for reply, I tried to do as you recommended, I moved
rewritehostport & force_send_socket into main routing block just before
route(1); I do understand that I'm doing two RRs, but here's TCPdump of the
softphone trying to Invite, notice that these are packets from OpenSIPS to
Asterisk, and they use Public IP to send SIP packets to internal Private IP:
00:10:05.942128 IP *<PUBLIC IP>*.5068 > 10.22.10.1.6000: UDP, length 961
.PV.k..PV.s...E..... at .@...w...
.
INVITE sip:555 at 10.22.10.1:6000 SIP/2.0
Record-Route: <sip:10.22.10.254:5068;r2=on;lr=on;ftag=79557f47>
Record-Route: <sip:*<PUBLIC IP>*:5066;r2=on;lr=on;ftag=79557f47>
Via: SIP/2.0/UDP 10.22.10.254:5068;branch=z9hG4bK8b37.90541a01.0
Via: SIP/2.0/UDP <PHONE IP>:6036;received=<PHONE
IP>;branch=z9hG4bK-d87543-fe60ef3b8e5cdd4d-1--d87543-;rport=6036
Max-Forwards: 69
Contact: <sip:6000@<PHONE IP>:6036>
To: <sip:555@*<PUBLIC IP>*:5066>
From: "User"<sip:6000@*<PUBLIC IP>*:5066>;tag=79557f47
Call-ID: 927d4958c229b528 at Zmxhc2g.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeam release 3015c stamp 27107
Content-Length: 190
v=0
o=- 3069147771 3069148115 IN IP4 <PHONE IP>
s=eyeBeam
c=IN IP4 <PHONE IP>
t=0 0
m=audio 8306 RTP/AVP 8 18 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
Then, Asterisk replies using private IPs:
00:10:05.942717 IP 10.22.10.1.6000 > 10.22.10.254.5068: UDP, length 637
.PV.s..PV.k...E...E... at .
.
.
.
.
..p....k7SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.22.10.254:5068;branch=z9hG4bK8b37.90541a01.0;received=*<PUBLIC
IP>*
Via: SIP/2.0/UDP <PHONE IP>:6036;received=<PHONE
IP>;branch=z9hG4bK-d87543-fe60ef3b8e5cdd4d-1--d87543-;rport=6036
From: "User"<sip:6000@*<PUBLIC IP>*:5066>;tag=79557f47
To: <sip:555@*<PUBLIC IP>*:5066>;tag=as0b8e2c25
Call-ID: 927d4958c229b528 at Zmxhc2g.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e41babb"
Content-Length: 0
And then OpenSIPS responds from Public IP again:
00:10:05.944903 IP *<PUBLIC IP>*.5068 > 10.22.10.1.6000: UDP, length 353
2010/12/7 Bogdan-Andrei Iancu <bogdan at voice-system.ro>
> Hi Yuri,
>
>
> Yuri Kirsanov wrote:
>
>> Hi community,
>>
>> I have following setup:
>>
>> Public IP (eth0) < OpenSIPS > Private IP (eth1) <------> Private IP (eth0)
>> Asterisk.
>>
>> I'm trying to relay all the initial packets received by OpenSIPS to
>> Asterisk, so I record-route initial packets and process all responses using
>> loose routing. OpenSIPS is restricted to use only two UDP IP addresses -
>> private & public ones. It all works fine on SIP layer, I do understand that
>> I also need RTP proxy, but at this stage I've got strange problem - with
>> mhomed=1 in configuration file OpenSIPS still sends packets using public IP
>> address, while Contact, To and Via fields are set correctly, to private IP
>> address of OpenSIPS.
>>
> OpenSIPS does not changes Contact and To IPs when forwarding a
> request.....are you sure you see such changes???
>
> The only IPs that reflects the used interface are the VIA and
> Record-Route...Actually in your case (if interface changing is done), you
> should have 2 RR headers...
>
> Also, using mhomed and force_send_socket() in the same time is a bit
> redundant - they do the same : setting the outgoing interface for the
> request:
> - mhomed is doing autodetection based on destination IP
> - force_send_socket() is setting the inteface from script.
>
> BTW, it is bogus to do:
> rewritehostport("10.22.10.1:6000 <http://10.22.10.1:6000/>");
>
> force_send_socket(10.22.10.254);
>
> in route[1] as this route is a generic relay route used also for sending
> sequential requests.....put that lines in main route just where you do
> record_route().
>
> Regards,
> Bogdan
>
>> I have tried to use force_send_socket but without any luck. Could you
>> please help me with this configuration? At least - why does OpenSIPS uses
>> public IP? Thanks!
>>
>> Here's my configuration:
>>
>> ####### Global Parameters #########
>> log_facility=LOG_LOCAL4
>> fork=yes
>> children=4
>> debug=9
>> disable_tcp=yes
>> auto_aliases=no
>> mhomed=1
>> port=5066
>> listen=udp:XXX.XXX.XXX.XXX:5066
>> listen=udp:10.22.10.254:5068 <http://10.22.10.254:5068/>
>>
>>
>> ####### Modules Section ########
>>
>> #set module path
>> mpath="/opt/opensips/lib64/opensips/modules/"
>>
>> loadmodule "signaling.so"
>> loadmodule "sl.so"
>> loadmodule "tm.so"
>> loadmodule "rr.so"
>> loadmodule "maxfwd.so"
>> loadmodule "textops.so"
>> loadmodule "mi_fifo.so"
>> loadmodule "uri.so"
>>
>> # ----------------- setting module-specific parameters ---------------
>>
>> # ----- mi_fifo params -----
>> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>>
>> # ----- rr params -----
>> # add value to ;lr param to cope with most of the UAs
>> modparam("rr", "enable_full_lr", 1)
>> modparam("rr", "append_fromtag", 1)
>>
>> # ----- uri params -----
>> modparam("uri", "use_uri_table", 0)
>>
>> ####### Routing Logic ########
>>
>> # main request routing logic
>>
>> route{
>>
>> if (!mf_process_maxfwd_header("10")) {
>> sl_send_reply("483","Too Many Hops");
>> exit;
>> }
>>
>> if (has_totag()) {
>> # sequential request withing a dialog should
>> # take the path determined by record-routing
>> if (loose_route()) {
>> if (is_method("BYE")) {
>> setflag(1); # do accounting ...
>> setflag(3); # ... even if the transaction
>> fails
>> } else if (is_method("INVITE")) {
>> # even if in most of the cases is useless,
>> do RR for
>> # re-INVITEs alos, as some buggy clients do
>> change route set
>> # during the dialog.
>> record_route();
>> }
>> # route it out to whatever destination was set by
>> loose_route()
>> # in $du (destination URI).
>> route(1);
>> } else {
>> if ( is_method("ACK") ) {
>> if ( t_check_trans() ) {
>> # non loose-route, but stateful
>> ACK; must be an ACK after
>> # a 487 or e.g. 404 from upstream
>> server
>> t_relay();
>> exit;
>> } else {
>> # ACK without matching transaction
>> ->
>> # ignore and discard
>> exit;
>> }
>> }
>> sl_send_reply("404","Not here");
>> }
>> exit;
>> }
>>
>> #initial requests
>>
>> # CANCEL processing
>> if (is_method("CANCEL"))
>> {
>> if (t_check_trans())
>> t_relay();
>> exit;
>> }
>>
>> t_check_trans();
>>
>> # preloaded route checking
>> if (loose_route()) {
>> xlog("L_ERR",
>> "Attempt to route with preloaded Route's
>> [$fu/$tu/$ru/$ci]");
>> if (!is_method("ACK"))
>> sl_send_reply("403","Preload Route denied");
>> exit;
>> }
>>
>> record_route();
>> route(1);
>> }
>>
>>
>> route[1] {
>> rewritehostport("10.22.10.1:6000 <http://10.22.10.1:6000/>");
>>
>> force_send_socket(10.22.10.254);
>> if (!t_relay()) {
>> sl_reply_error();
>> };
>> exit;
>> }
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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