[Users] WeSIP setup problem

[GP.Coder]

Hi,Gines
	Could you send me your configuration files of the 3 mode ?  
Thanks  

GP.Coder
2007-04-28



·¢¼þÈË£º Ginés_Gómez
·¢ËÍʱ¼ä£º 2007-04-27 20:52:22
ÊÕ¼þÈË£º ventilo35 at free.fr
³­ËÍ£º users openser.org
Ö÷Ì⣺ Re: [Users] WeSIP setup problem

Hi ,

I've discovered that there is a problem with WeSIP looping back to  
its own address (which is what you are trying to do). We are fixing  
it but it'll take more time. As a workaround you can do thre  
different things.

1) Let OpenSER discover the next hop before passing the message to  
WeSIP then proxy from WeSIP to the next hop.
<--------- Transaction 1 --------- >   <-------- Transaction  
2------------ >
UAC --------- >  [OpenSER -- > WeSIP -- > OpenSER] --------- > UAS
1.1 UAC sends request with RURI=sip:5002 at 192.168.1.11
1.2 OpenSER modifies the RURI after consulting the location info   
RURI=sip:5002 at 192.168.1.16
1.3 OpenSER passes the modified message to WeSIP
1.4 WeSIP processes and proxies
1.5 OpenSER processes message coming from WeSIP (if X-WeSIP-SPIRAL  
was used) and forwards it downstream

2) Use separated WeSIP with its own OpenSER frontend( see Openser.cfg  
working as WeSIP front-end at http://www.wesip.com/mediawiki/ 
index.php/Configuration#Openser.cfg_in_standalone)
<---Trans 1---- > <----- Transaction 2 ------ > <-----Transaction 3------ 
  > <---Trans 4--- >

UAC ---- > OpenSER ---- > (WeSIP with OpenSER frontend) ---- > OpenSER  
------ > UAS

2.1 UAC sends request with RURI=sip:5002 at 192.168.1.11
2.2 OpenSER delivers message to standalone WeSIP
2.3 WeSIP processes and proxies back to OpenSER (WeSIP may  
RecordRoute or not)
2.4 OpenSER modifies the RURI after consulting the location info   
RURI=sip:5002 at 192.168.1.16
2.5 OpenSER proxies to UAS

3)     Use different IP to make WeSIP think that it is forwarding to  
a different host and not looping

<-------Transaction 1-------- > <------------- Transaction 2  
------------- > <---Transaction 3---- >

UAC ---- >  [OpenSER -- > WeSIP -- > OpenSER] --------- > OpenSER  
(192.168.1.12) ------ > UAS
1.1 UAC sends request with RURI=sip:5002 at 192.168.1.11
1.2 OpenSER modifies the RURI to an alias IP RURI=sip: 
5002 at 192.168.1.12 (192.168.1.12 is a second IP where OpenSER listens)
1.3 OpenSER passes the modified message to WeSIP
1.4 WeSIP processes and proxies to 192.168.1.12 (No loopback failure  
since 192.168.1.12 != 192.168.1.11)
1.5 OpenSER processes message coming from WeSIP (if X-WeSIP-SPIRAL  
was used) and forwards it to 192.168.1.12
1.6 OpenSER receives proxied message (in IP 192.168.69.12)
1.7 OpenSER applies location translation
1.8 OpenSER forwards


Hope it helps

Regards

Gines


> Hi Gines, thanks for your support
>
>
> > Your servlet named "neo" is being executed and looks like it is
> > proxying but what from what I see in the packet it is proxying to
> > itself. Could you try to proxy to another address and see if this
> > still happens ? Aside, could you send me your configuration files and
> > your servlet code ?
>
> The serlvet code is the following
>
> protected void doInvite(SipServletRequest callerInvite) throws  
> ServletException,
> IOException {
>
>
>       //Get a proxy for this request
>       Proxy callerProxy = callerInvite.getProxy();
>
>  System.out.println("doInvite within neos servlet: requestURI is " +
> callerInvite.getRequestURI() );
>
>
>  callerProxy.proxyTo(callerInvite.getRequestURI());
>
>
>     }
>
>
> This code causes the "no route" exception. The println above gives :
>
> doInvite within neos servlet: requestURI is sip:5002 at 192.168.1.11
>
> where 5002 is the called SIP phone and 192.168.1.11 is the address  
> of my OpenSER
> + WeSIP box (i.e. the managed SIP domain)
>
>
> Following your advice I have tried to change the proxyTo to a  
> different address:
>
>
>  SipURI dest = sipFactory.createSipURI("5002","192.168.1.16");
>       callerProxy.proxyTo(dest);
>
> It that case, the called phone (5002, IP address 192.168.1.16) gets  
> the INVITE
> correctly. However, something puzzles me here: the serlvet is not  
> supposed to
> know the IP address of the called phone. In my understanding, the  
> AS processes
> the INVITE, then it must send it back to the proxy server which is  
> supposed to
> forward it to the next hop, is that right ?
>
> Could you please give me an example of a doInvite which basically  
> does nothing
> and which lets the INVITE be forwarded to the called party ?
>
>
> Finally here are my config files:
>
>  < < < < < < < < < Server.xml  > > > > > > > > >
>
>  <Server port="5005" shutdown="SHUTDOWN" debug="1" >
>  <Service name="NEOS-Standalone" >
>       <Connector  
> className="org.apache.catalina.connector.http.HttpConnector"
>    port="7080" minProcessors="5" maxProcessors="5"
>    enableLookups="true" acceptCount="10" debug="0" / >
>
>  <Connector className="com.voztele.sipservlet.connector.SipConnector"
>    minProcessors="5" maxProcessors="5"
>    enableLookups="true" acceptCount="10" debug="1"
>    addresses="192.168.1.11:5060" pathName="com.voztele"  
> protocol="udp"  >
>    <ExtraProperties >
>    <Property key="com.voztele.javax.sip.SER_ADDRESS"  
> value="127.0.0.1" / >
>    <Property key="com.voztele.javax.sip.SER_PORT" value="5080" / >
>    <Property key="com.voztele.STATISTICS" value="false" / >
>  <Property key="javax.sip.STACK_NAME" value="neos_as" / >
>  <Property key="javax.sip.THREAD_POOL_SIZE" value="10" / >
>  <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE"  
> value="10" / >
>  <Property key="com.voztele.javax.sip.SPIRAL_HDR" value="true" / >
>  <Property key="com.voztele.javax.sip.REENTRANT_LISTENER"  
> value="true" / >
>  <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT"  
> value="false" / >
>  <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="3" / >
>    </ExtraProperties >
>    </Connector >
>
>       <Engine name="Standalone" defaultHost="localhost" debug="1"
> className="com.voztele.sipservlet.core.ConvergedEngine" >
>
>  <Mapper className="org.apache.catalina.core.StandardEngineMapper"
> protocol="HTTP/1.1"/ >
>  <Mapper className="org.apache.catalina.core.StandardEngineMapper"
> protocol="HTTP/1.0"/ >
>    <Mapper className="com.voztele.sipservlet.core.EngineSipMapper"
> protocol="SIP/2.0"/ >
>  <Logger className="org.apache.catalina.logger.FileLogger"  
> timestamp="true"/ >
>  <Realm className="org.apache.catalina.realm.MemoryRealm" / >
>
>  <Host name="localhost" nameSip="192.168.1.11_5060" debug="1"
> appBase="wesipapps"  
> className="com.voztele.sipservlet.core.ConvergedHost" >
>
>  <Logger className="org.apache.catalina.logger.FileLogger"  
> timestamp="true"/ >
>  <Mapper className="com.voztele.sipservlet.core.SipHostMapper"
> protocol="SIP/2.0"/ >
>  <Mapper className="org.apache.catalina.core.HttpHostMapper"
> protocol="HTTP/1.1"/ >
>  <Mapper className="org.apache.catalina.core.HttpHostMapper"
> protocol="HTTP/1.0"/ >
>       </Host >
>     </Engine >
>  </Service >
>  </Server >
>
>
>  < < < < < < < < < <openser.cfg > > > > > > > > > > > > > >
> #
> # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $
> #
> # simple quick-start config script
> # Please refer to the Core CookBook at http://www.openser.org/ 
> dokuwiki/doku.php
> # for a explanation of possible statements, functions and parameters.
> #
>
> # ----------- global configuration parameters ------------------------
>
> debug=3            # debug level (cmd line: -dddddddddd)
> fork=yes
> #log_stderror=no    # (cmd line: -E)
> children=4
>
> # Uncomment these lines to enter debugging mode
> #fork=no
> log_stderror=yes
> #
>
>
> listen = tcp:192.168.1.11:5060
> listen = udp:192.168.1.11:5060
>
>
> # ------------------ module loading ----------------------------------
>
> #set module path
> mpath="/usr/local/lib/openser/modules/"
>
> # Uncomment this if you want to use SQL database
> #loadmodule "mysql.so"
>
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "rr.so"
> loadmodule "maxfwd.so"
> loadmodule "usrloc.so"
> loadmodule "registrar.so"
> loadmodule "textops.so"
> loadmodule "mi_fifo.so"
>
> # added for WeSIP
> loadmodule "seas.so"
> modparam("seas","listen_sockets","127.0.0.1:5080")
> # end added for WeSIP
>
> # Uncomment this if you want digest authentication
> # mysql.so must be loaded !
> #loadmodule "auth.so"
> #loadmodule "auth_db.so"
>
> # ----------------- setting module-specific parameters ---------------
>
> # -- mi_fifo params --
>
> modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
>
> # -- usrloc params --
>
> modparam("usrloc", "db_mode",   0)
>
> # Uncomment this if you want to use SQL database
> # for persistent storage and comment the previous line
> #modparam("usrloc", "db_mode", 2)
>
> # -- auth params --
> # Uncomment if you are using auth module
> #
> #modparam("auth_db", "calculate_ha1", yes)
> #
> # If you set "calculate_ha1" parameter to yes (which true in this  
> config),
> # uncomment also the following parameter)
> #
> #modparam("auth_db", "password_column", "password")
>
> # -- rr params --
> # add value to ;lr param to make some broken UAs happy
> modparam("rr", "enable_full_lr", 1)
>
> # -------------------------  request routing logic -------------------
>
> # main routing logic
>
> route{
>
>  # initial sanity checks -- messages with
>  # max_forwards==0, or excessively long requests
>  if (!mf_process_maxfwd_header("10")) {
>  sl_send_reply("483","Too Many Hops");
>  exit;
>  };
>
>  if (msg:len  >=  2048 ) {
>  sl_send_reply("513", "Message too big");
>  exit;
>  };
>
>  # we record-route all messages -- to make sure that
>  # subsequent messages will go through our proxy; that's
>  # particularly good if upstream and downstream entities
>  # use different transport protocol
>  if (!method=="REGISTER")
>  record_route(); # ftoutain essai de supprimer ça ? ne change rien
>
>  # subsequent messages withing a dialog should take the
>  # path determined by record-routing
>  if (loose_route()) {
>  # mark routing logic in request
>  append_hf("P-hint: rr-enforced\r\n");
>  route(1);
>  };
>
>  if (!uri==myself) {
>  # mark routing logic in request
>  append_hf("P-hint: outbound\r\n");
>
>  route(1);
>  };
>
>  # if the request is for other domain use UsrLoc
>  # (in case, it does not work, use the following command
>  # with proper names and addresses in it)
>  if (uri==myself) {
>
>  if (method=="REGISTER") {
>
>  # Uncomment this if you want to use digest authentication
>  #if (!www_authorize("openser.org", "subscriber")) {
>  # www_challenge("openser.org", "0");
>  # exit;
>  #};
>
>  save("location");
>  exit;
>  };
>
>  lookup("aliases");
>  if (!uri==myself) {
>  append_hf("P-hint: outbound alias\r\n");
>  route(1);
>  };
>
>  # native SIP destinations are handled using our USRLOC DB
>  if (!lookup("location")) {
>  sl_send_reply("404", "Not Found");
>  exit;
>  };
>  append_hf("P-hint: usrloc applied\r\n");
>  };
>
>  route(1);
> }
>
>
> route[1] {
>
>  if (is_present_hf("X-WeSIP-SPIRAL")) {
>  if (!t_relay()) {
>  sl_reply_error();
>  };
>  exit;
>  };
>
>  if (! as_relay_t("neos_as") ){
>  t_reply("500", "NEOS service not found");
>  };
>
>  exit;
> }
>
>
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