[Users] Mediaproxy newbie
Daniel-Constantin Mierla
daniel at voice-system.ro
Thu Oct 19 22:40:52 CEST 2006
Hello,
maybe this link helps:
http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy
Cheers,
Daniel
On 10/18/06 19:04, Marnus van Niekerk wrote:
> Hi,
>
> I am trying to set up openser with mediaproxy (at xx.xx.xx.133) to
> route calls from UA behind NAT to asterisk as voicemail (at
> xx.xx.xx.134) and PSTN gateways (at xx.xx.xx.32)
>
> I can see in the SDP payload that the RTP is being sent from asterisk
> to mediaproxy, but in sessions.py it shows the private ip not the
> public one and I have one way audio.
>
> Can anybody help please.
>
> opnser.cfg below.
>
> Marnus
>
> --
>
> debug=3 # debug level (cmd line: -dddddddddd)
> fork=yes
> log_stderror=no # (cmd line: -E)
> log_facility=LOG_LOCAL6
>
> check_via=no # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
> port=5060
> children=4
> fifo="/tmp/openser_fifo"
>
> # ------------------ module loading ----------------------------------
> loadmodule "/usr/local/lib/openser/modules/mysql.so"
> loadmodule "/usr/local/lib/openser/modules/sl.so"
> loadmodule "/usr/local/lib/openser/modules/tm.so"
> loadmodule "/usr/local/lib/openser/modules/rr.so"
> loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
> loadmodule "/usr/local/lib/openser/modules/usrloc.so"
> loadmodule "/usr/local/lib/openser/modules/registrar.so"
> loadmodule "/usr/local/lib/openser/modules/textops.so"
> loadmodule "/usr/local/lib/openser/modules/uri_db.so"
> loadmodule "/usr/local/lib/openser/modules/domain.so"
> loadmodule "/usr/local/lib/openser/modules/mediaproxy.so"
> loadmodule "/usr/local/lib/openser/modules/nathelper.so"
>
> # Logging
> loadmodule "/usr/local/lib/openser/modules/xlog.so"
>
> loadmodule "/usr/local/lib/openser/modules/auth.so"
> loadmodule "/usr/local/lib/openser/modules/auth_db.so"
>
> # ----------------- setting module-specific parameters ---------------
>
> # -- usrloc params --
> modparam("usrloc", "db_mode", 0)
>
> modparam("usrloc", "db_mode", 2)
>
> modparam("auth_db", "calculate_ha1", yes)
> modparam("auth_db", "password_column", "password")
>
> modparam("rr", "enable_full_lr", 1)
>
> #tm timeout for voicemail params
> modparam("tm", "fr_timer", 3)
> modparam("tm", "fr_inv_timer", 35)
> modparam("tm", "noisy_ctimer", 1)
>
> # parms for NAT/mediaproxy
> modparam("nathelper", "rtpproxy_disable", 1)
> modparam("nathelper", "natping_interval", 0)
> modparam("mediaproxy", "natping_interval", 30)
> modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock")
> modparam("mediaproxy", "sip_asymmetrics",
> "/usr/local/etc/openser/sip-clients")
> modparam("mediaproxy", "rtp_asymmetrics",
> "/usr/local/etc/openser/rtp-clients")
> modparam("registrar", "nat_flag", 6)
>
>
> # ------------------------- request routing logic -------------------
>
> # main routing logic
> route {
> #check for old messages: could mean a problem with the DNS
> entries or some other loop-causer...
> if (!mf_process_maxfwd_header("10"))
> {
> xlog("L_WARN", "WARNING: Too many hops\n");
> sl_send_reply("483", "Too many hops, forward count
> exceeded limit\n");
> return;
> };
>
> #check for extremely large messages; we don't need a sip dos
> attack
> if (msg:len >= 2048)
> {
> xlog("L_WARN", "WARNING: Message too large, &>= 2048
> bytes\n");
> sl_send_reply("513", "Message too large, exceeded
> limit\n");
> return;
> };
>
> # Track what is happening
> xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n");
>
> #record everything besides registers and acks
> if(method!="REGISTER" && method!="ACK")
> {
> setflag(1);
> };
>
> # Record Route Section
> if (method=="INVITE" && client_nat_test("3"))
> {
> record_route_preset("xx.xx.xx.133:5060;nat=yes");
> }
> else if (method!="REGISTER")
> {
> record_route();
> }
>
> # Call tear down section
> if (method=="BYE" || method=="CANCEL")
> {
> end_media_session();
> }
>
> #do not send to voicemail if BYE or CANCEL
> #is used to end call before user pickup or timeout
> if(method=="CANCEL" || method=="BYE")
> {
> setflag(10);
> };
>
> #grant route if route headers already present
> if (loose_route())
> {
> # May need client_nat_test & use_media_proxy here...
> route(1);
> return;
> };
>
> #Always require authentication, which could result in a PSTN
> if (method=="REGISTER")
> {
>
> if (!search("^Contact:[ ]*\*") && client_nat_test("7"))
> {
> setflag(6);
> fix_nated_register();
> force_rport();
> };
>
> if(!www_authorize("domain.tld", "subscriber"))
> {
> www_challenge("domain.tld", "0");
> return;
> }
> else
> {
> if (!check_to())
> {
> sl_send_reply("401", "Unauthorized");
> return;
> };
>
> #Save into user database, used below when
> checking if user is available
> xlog("L_INFO", "REGISTER: User $fu
> Authenticated Correctly\n");
> save("location");
> return;
> };
> };
>
> if (method=="INVITE")
> {
> if (client_nat_test("3"))
> {
> setflag(7);
> force_rport();
> fix_nated_contact();
> };
>
> if(uri=~"sip:\*86 at .*")
> {
> #authorize if a call is going to VM
> if(!proxy_authorize("domain.tld", "subscriber"))
> {
> proxy_challenge("domain.tld", "0");
> return;
> };
>
> xlog("L_INFO", "CALL: Call from $fu to check
> voicemail\n");
> rewritehostport("vm.domain.tld:5060");
> }
> else
> {
> if (does_uri_exist())
> {
> #Call is to sip client, so do nothing
> but route
> xlog("L_INFO", "CALL: Sip client\n");
> if (!lookup("location"))
> {
> sl_send_reply("404", "Not
> Found");
> xlog("L_ERROR", "ERROR: User
> $tu Not Found\n");
> return;
> };
> }
> else
> {
> #authorize if a call is going to PSTN
> if(!proxy_authorize("domain.tld",
> "subscriber"))
> {
> proxy_challenge("domain.tld",
> "0");
> return;
> };
>
> #Call destination is PSTN, so send it
> to the gateway
> xlog("L_INFO", "CALL: PSTN $tu from $fu
> \n");
> rewritehostport("ast1.domain.tld:5060");
> };
> };
>
> #Make sure that all subsequent requests go through us;
> #done at the top already
> #record_route();
> }
> else
> {
> if (does_uri_exist())
> {
> #Call is to sip client, so do nothing but route
> xlog("L_INFO", "CALL: Sip client\n");
> if (!lookup("location"))
> {
> sl_send_reply("404", "Not Found");
> xlog("L_ERROR", "ERROR: User $tu Not
> Found\n");
> return;
> };
> }
> else
> {
> #Call destination is PSTN, so send it to the
> gateway
> xlog("L_INFO", "CALL: PSTN $tu from $fu \n");
> rewritehostport("ast1.domain.tld:5060");
> };
> #record_route();
> };
>
> #ALL PROCESSING IS DONE, SO ROUTE
> route(4);
> route(1);
> }
>
> route[1]
> {
> #send the call outward
> if(method=="INVITE" && !isflagset(10))
> {
> t_on_failure("2"); # voicemail if failure
> };
>
> if (!t_relay())
> {
> xlog("L_WARN", "ERROR: t_relay failed");
> sl_reply_error();
> };
> }
>
> # -----------------------------------------------------------------
> # NAT Traversal Section
> # -----------------------------------------------------------------
> route[4]
> {
> if (isflagset(6) || isflagset(7))
> {
> if (!isflagset(8))
> {
> setflag(8);
> use_media_proxy();
> };
> };
> }
>
> failure_route[2]
> {
> if(!t_was_cancelled() && !t_check_status("407"))
> {
> revert_uri();
> rewritehostport("vm.domain.tld:5060");
> append_branch();
> #PREVENT SOME CRAZY VOICEMAIL LOOP
> xlog("L_INFO", "INFO: CALL TO VOICEMAIL");
> setflag(10);
> route(1);
> }
> }
>
> onreply_route[1]
> {
> if ((isflagset(6) || isflagset(7)) &&
> (status=~"(180)|(183)|2[0-9][0-9]"))
> {
> if (!search("^Content-Length:[ ]*0"))
> {
> use_media_proxy();
> };
> };
>
> if (client_nat_test("1"))
> {
> fix_nated_contact();
> };
> }
>
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