sr-users November 2004

sr-users@lists.kamailio.org

203 participants
386 discussions

[Serusers] RTP Proxy: can' t have RTP traversing NAT

by frbilles＠libertysurf.fr

Hi. I've been trying to fix this issue by myself for about a month but definitely needs your help. I use SER 0.8.14 from CVS sources with RTPProxy V 1.19 and can't have RTP stream working through NAT (the phone rings -- SIP is OK). SER and RTPProxy run on the same server. RTPProxy is up and running, and I set 777 rights to the socket: [root@servername rtpproxy]# ll /var/run/rtpproxy.sock srwxrwxrwx 1 root root 0 Nov 8 16:22 /var/run/rtpproxy.sock= I use as a client X-Lite V2.0 on each side. Please find below the log (call initiated from Internet to a callee on our LAN) and the excellent ser.cfg file I found in the Serusers Archives. Thank you in advance for your help. Francois. ===================================================================================================================== LOG: ===================================================================================================================== Maxfwd module- initializing 0(28184) mod_init(): Database connection opened successfuly textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 1(28186) rtpp_test: RTP proxy found, support for it enabled 5(28195) INFO: fifo process starting: 28195 2(28187) rtpp_test: RTP proxy found, support for it enabled 3(28188) rtpp_test: RTP proxy found, support for it enabled 4(28194) rtpp_test: RTP proxy found, support for it enabled 5(28195) rtpp_test: RTP proxy found, support for it enabled 5(28195) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 8(28208) rtpp_test: RTP proxy found, support for it enabled 6(28206) rtpp_test: RTP proxy found, support for it enabled 10(28213) rtpp_test: RTP proxy found, support for it enabled 7(28207) rtpp_test: RTP proxy found, support for it enabled 11(28214) rtpp_test: RTP proxy found, support for it enabled 9(28209) rtpp_test: RTP proxy found, support for it enabled 0(28184) rtpp_test: RTP proxy found, support for it enabled 4(28194) ------------------------------------------- 4(28194) entering main loop 4(28194) src address different than via header->NAT detected 4(28194) force_rport and fix_nated_contact and setflag(5) 4(28194) INVITE message received 4(28194) ------------------------------------------- 4(28194) entering route[1] - relaying SIP message 4(28194) at least one of the participants is NATed->record_route 4(28194) -->setting up reply processing ->onreply_route[1] 4(28194) INVITE request-->force_rtp_proxy, set NATED-INVITE flag(7) 4(28194) relaying message ... 3(28188) ------------------------------------------- 3(28188) onreply_route[1] entered 3(28188) status 100 received 4(28194) ------------------------------------------- 4(28194) onreply_route[1] entered 4(28194) status 180 received 2(28187) ------------------------------------------- 2(28187) onreply_route[1] entered 2(28187) status 2xx or 183 2(28187) marked(7) as NATED-INVITE -> force_rtp_proxy 2(28187) ERROR: send_rtpp_command: can't read reply from a RTP proxy 2(28187) ------------------------------------------- 2(28187) onreply_route[1] entered 2(28187) status 2xx or 183 2(28187) marked(7) as NATED-INVITE -> force_rtp_proxy 2(28187) ERROR: send_rtpp_command: can't connect to RTP proxy 3(28188) ------------------------------------------- 3(28188) entering main loop 3(28188) BYE message received 3(28188) ------------------------------------------- ======================================================================================================================== SER.CFG ======================================================================================================================== # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=<ip address in the DMZ> #listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=servername.mycompany.com alias=mycompany.com localhost # Uncomment these lines to enter debugging mode #debug=7 #fork=no #log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # load the voicemail module #loadmodule "/usr/local/lib/ser/modules/vm.so" # load the enum module loadmodule "/usr/local/lib/ser/modules/enum.so" # load the group module, to verify if a user forwards to voicemail loadmodule "/usr/local/lib/ser/modules/group.so" # load the nathelper module loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- registrar parameter # special NAT flag indicates that a registered client is behind NAT modparam("registrar", "nat_flag", 6) # -- 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) #modparam("usrloc", "db_url", "mysql://login:password@localhost/ser") modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://login:password@localhost/ser") # -- 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") #modparam("auth_db", "db_url", "mysql://login:password@localhost/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- voicemail params -- #modparam("voicemail", "db_url","mysql://login:password@localhost/ser") # -- voicemail params -- #modparam("group", "db_url","mysql://login:password@localhost/ser") # -- nathelper params -- modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 30 ) #modparam("tm", "fr_inv_timer", 8 ) #Explicitly set the socket used by rtpproxy #modparam("nathelpler", "rtpproxy_sock", "/var/run/rtpproxy.sock") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); if (nat_uac_test("2")) { log(1, "src address different than via header->NAT detected\n"); log(1, "force_rport and fix_nated_contact and setflag(5)\n"); #try NAT traversal, works only if the client is symmetrical force_rport(); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); # flag 5 indicates that incoming request is from NATed client setflag(5); }; if (method=="REGISTER") log(1, "REGISTER message received\n"); if (method=="INVITE") log(1, "INVITE message received\n"); if (method=="ACK") log(1, "ACK message received\n"); if (method=="BYE") log(1, "BYE message received\n"); if (method=="CANCEL") log(1, "CANCEL message received\n"); if (method=="SUBSCRIBE") log(1, "SUBSCRIBE message received\n"); if (method=="NOTIFY") log(1, "NOTIFY message received\n"); if (method=="OPTIONS") log(1, "OPTIONS message received\n"); if (method=="INFO") log(1, "INFO message received\n"); if (method=="MESSAGE") log(1, "MESSAGE message received\n"); if (method=="REFER") log(1, "REFER message received\n"); # 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"); break; }; if (msg:len > max_len) { #if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # loose-route processing if (loose_route()) { log(1, "loose_route processing\n"); t_relay(); break; }; # create transaction state; abort if error occured # if ( !t_newtran()) { # sl_reply_error(); # break; # }; #new # now check if it's about PSTN destinations through our gateway; # note that 8.... is exempted for numerical non-gw destinations if (uri=~"^sip:0[0-9]*@.*") { route(3); break; }; # # 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") { log(1, "analyzing REGISTER request\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("servername.mycompany.com", "subscriber")) { www_challenge("servername.mycompany.com", "0"); break; }; if (isflagset(5)) { #register from nated client, save nat_flag=6 #in location table setflag(6); }; if (!save("location")) { log(1, "save location error\n"); sl_reply_error(); }; break; }; lookup("aliases"); #mark transaction for voicemail if (is_user_in("Request-URI", "voicemail\n")) { log(1, "requested user is in voicemail group"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found log(1, "requested user not found\n"); route(4); break; }; }; #add failure route which should be performed if response code >=300 if (method=="INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP route(1); } route[1]{ log(1, "-------------------------------------------\n"); log(1, "entering route[1] - relaying SIP message\n"); if ((isflagset(5)) || (isflagset(6))) { log(1, "at least one of the participants is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy, set NATED-INVITE flag(7)"); force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; }; log(1, "relaying message ...\n"); if (!t_relay()) { log(1, "t_relay error occured\n"); sl_reply_error(); }; } # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { log(1, "-------------------------------------------\n"); log(1, "onreply_route[1] entered\n"); if (isflagset(6)) { log(1, "transaction was sent to a NATED client -> fix nated contact\n"); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for response\r\n"); } if ( (status=~"100") ) { log(1, "status 100 received\n"); }; if ( (status=~"180") ) { log(1, "status 180 received\n"); }; if ( (status=~"202") ) { log(1, "status 202 received\n"); }; if ( (status=~"200" || status=~"183") ) { log(1, "status 2xx or 183"); if ( isflagset(7) ) { log(1, "marked(7) as NATED-INVITE -> force_rtp_proxy \n"); force_rtp_proxy(); append_hf("P-hint: response forced to rtp proxy\r\n"); }; }; } #new # logic for calls to the PSTN route[3] { # turn accounting on setflag(1); /* require all who call PSTN to be members of the "int" group; apply ACLs only to INVITEs -- we don't need to protect other requests, as they don't imply charges; also it could cause troubles when a call comes in via PSTN and goes to a party that can't authenticate (voicemail, other domain) -- BYEs would fail then; exempt Cisco gateway from authentication by IP address -- it does not support digest */ if (method=="INVITE" && (!src_ip==WhateverIP)) { if (!proxy_authorize( "servername.mycompany.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "servername.mycompany.com" /* realm */, "0" /* no qop */ ); break; }; # let's check from=id ... avoids accounting confusion if(!is_user_in("credentials", "int")) { sl_send_reply("403", "NO PSTN Privileges..."); break; }; consume_credentials(); }; # INVITE to authorized PSTN # if you have passed through all the checks, let your call go to GW! force_rtp_proxy(); record_route(); t_on_reply("1"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("212.17.35.184","5060"); } route[4]{ log(1, "-------------------------------------------\n"); log(1, "entering route[4] = requested user not online\n"); # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "REFER" || method == "BYE")) { log(1, "no invite,ack,cancel,refer->return 404\n"); sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber and no echo/conference call if ( isflagset(4)) { log(1, "flag(4) active\n"); }; if (uri =~ "conference") { log(1, "conference call\n"); }; if (uri =~ "echo") { log(1, "echo call\n"); }; if ( !( isflagset(4) || (uri =~ "conference") || (uri =~ "echo") ) ) { log(1, "no voicemail subscriber->return 404"); sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; if ( isflagset(5) ) { log(1, "caller is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy"); force_rtp_proxy(); }; }; # forward to voicemail now ² rewritehostport("WhateverIP:5060"); log(1, "forward to voicemail\n"); t_relay_to_udp("WhateverIP", "5060"); } failure_route[1] { /* XX: note: unsafe if preloaded routes without username used */ log(1, "-------------------------------------------\n"); log(1, "failureroute[1] entered\"); revert_uri(); rewritehostport("WhateverIP:5060"); append_branch(); t_relay_to_udp("WhateverIP", "5060"); } ************************ ADSL ILLIMITE TISCALI + TELEPHONE GRATUIT ************************ Surfez 40 fois plus vite pour 30EUR/mois seulement ! Et téléphonez partout en France gratuitement, vers les postes fixes (hors numéros spéciaux). Tarifs très avantageux vers les mobiles et l'international ! Pour profiter de cette offre exceptionnelle, cliquez ici : http://register.tiscali.fr/adsl (voir conditions sur le site)

19 years, 6 months

[Serusers] Sip-Username and Sip-Password

by Ricardo Martinez

Hello List. I have a question regarding to the User and Password configured in a UA. In which part of all the authentication process the password is used by SER? If i have Radius for authenticate users, this password is "encrypted" in the nonce parameter? If so, what prevent for some attacker to "intercept" this packet and obtain this values?. Thanks in advance Ricardo Martinez

19 years, 6 months

[Serusers] RTP Proxy: can' t have RTP traversing NAT

by frbilles＠libertysurf.fr

19 years, 6 months

[Serusers] RTP Proxy: can' t have RTP traversing NAT

by frbilles＠libertysurf.fr

19 years, 6 months

[Serusers] rtpproxy+nathelper (0.8.14) +video

by blanca_mayayo＠lycos.es

19 years, 6 months

[Serusers] error restarting ser

by Vinayak Krishan

I'm installing SER 8.12 on a Fedora Box with MySQL 4.1.7-0... I've followed the steps in SER-howto completely until I get to the section down the middle where it asks to modify the ser.cfg file.. After modifying it for mysql, digest authentication and changing the uri to uri=~"vinayakk.local" , the ser restart command comes back with a "error in config file" message .. Could you let me know how I can find out what is causing this error? thanks

19 years, 6 months

[Serusers] HELP!!! Problem with registration

by mahesh

Hi All, Could someone please provide some information about the following issue. To be more specific on the problem, I am using the following network configuration. HOST1 XP---FW/NAT---Internet---HOST2 XP Network Configuraton: Host1's IP address : 172.16.4.8 FW Internal address: 172.16.4.1 FW external address: 202.125.84.163 Host2's IP address : 202.125.84.164 SER(public server) : 195.37.77.99 FW/NAT has SIP-ALG implementation. SJPhone is installed on both Host1 and Host2. Now Host1 registers with the SIP proxy server. The register request after the NAT(ALG) process has the src port of 55001(IP:202.125.84.163) and the contact port of 60002(IP:202.125.84.163). Similarly SJphone on Host2 is also registered with the proxy server. Now when a call is made to Host1 from Host2 using SJPhone, the SIP invite request coming from Host2 is sent to the proxy server, which in turn is sending it to 202.125.84.163 on port 55001(source port of register). This request is supposed to come on port 60002(contact port of register). A month back when i tested in the same configuration, the request from the external machine was coming on the contact port of the register request(according to the RFC 3261). But when i tested it yesterday it was having the mentioned problem behavior. Could someone please let me know if anything is changed in the proxy server(SER) which is causing this behavior. Also how this behavior can be avoided. Thanks in advance. Best Regards Mahesh

19 years, 6 months

[Serusers] Having trouble with SER Web

by Bobby Lloyd

OK, it’s all installed and running, but when I try to log in it just goes back to the log in screen, no errors, no warnings, nothing. I know I’m missing something very basic here. Any ideas? Thanks in advance, Bobby --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.786 / Virus Database: 532 - Release Date: 10/29/2004

19 years, 6 months

[Serusers] New SER release

by Bogdan-Andrei IANCU

Hi everybody, a new SER release was decided to come out (Christmas is coming ;-)). It will be based on the current unstable version, so we can say it will bring radical changes. Probably to get it a stable phase will take some time and effort. As a first step we will branch the current CVS head and the commits will be stopped - excepting bug fixes. Of course, the CVS head will go further normally. The branch it will be done on 17th 24:00. So, if there are any pending commits that should go into release, please do them till 17th. Once the branch is created, I will try to compile a list with all important changes and based on this, we will get into testing phase - any external help will be strongly appreciated. regards, bogdan

19 years, 6 months

[Serusers] Newbie Question ...

by Dana Rossetti

Hello All, Can anyone offer an opinion on which PC Linux distribution/hardware platform works well with SIP Express Router ? Thanks in advance. Dana Rossetti drossetti(a)gmail.com

19 years, 6 months

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sr-users November 2004