Edoardo Serra wrote:
At 13.51 19/12/2006, Klaus Darilion wrote:
You said that the 200 contains openser's IP in the SDP? Is it put in there by openser or already by Asterisk?
Tnx very much for help
It's put in there by OpenSER.
I'm attaching the 2 SIP/SDP packets (1 from asterisk to openser and 1 from openser to client)
AAA.AAA.AAA.AAA stands for IP of Asterisk OOO.OOO.OOO.OOO stands for IP of OpenSER CCC.CCC.CCC.CCC stands for IP of client 3333333333 is the called number
No. Time Source Destination Protocol Info 20 12.646925 AAA.AAA.AAA.AAA OOO.OOO.OOO.OOO SIP/SDP Status: 200 OK, with session description
Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP OOO.OOO.OOO.OOO;branch=z9hG4bK5bbd.eaf4f093.0;received=OOO.OOO.OOO.OOO Via: SIP/2.0/UDP CCC.CCC.CCC.CCC:8952;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
Record-Route: <sip:OOO.OOO.OOO.OOO;lr=on;ftag=9043ec70> From: "test"<sip:test@OOO.OOO.OOO.OOO>;tag=9043ec70 To: "3333333333"<sip:3333333333@OOO.OOO.OOO.OOO>;tag=as30a7528b Call-ID:98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU. CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:3333333333@AAA.AAA.AAA.AAA:5060 Content-Type: application/sdp Content-Length: 291 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 20137 20138 IN IP4 AAA.AAA.AAA.AAA Session Name (s): session Connection Information (c): IN IP4 AAA.AAA.AAA.AAA Time Description, active time (t): 0 0 Media Description, name and address (m): audio 58508 RTP/AVP 98 3 8 0 101 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - -
No. Time Source Destination Protocol Info 21 12.647437 OOO.OOO.OOO.OOO CCC.CCC.CCC.CCC SIP/SDP Status: 200 OK, with session description
Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
^^^^^^^^^here should be CCC.CCC.CCC.CCC
maybe a search/replace typo?
have you sniffed directly on the SIP proxy server? (Maybe is there a SIP ALG somewhere inbetween)
regards klaus
Record-Route: <sip:OOO.OOO.OOO.OOO;lr=on;ftag=9043ec70> From: "test"<sip:test@OOO.OOO.OOO.OOO>;tag=9043ec70 To: "3333333333"<sip:3333333333@OOO.OOO.OOO.OOO>;tag=as30a7528b Call-ID:98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU. CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:3333333333@AAA.AAA.AAA.AAA:5060 Content-Type: application/sdp Content-Length: 291 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 20137 20138 IN IP4 OOO.OOO.OOO.OOO Session Name (s): session Connection Information (c): IN IP4 OOO.OOO.OOO.OOO Time Description, active time (t): 0 0 Media Description, name and address (m): audio 58508 RTP/AVP 98 3 8 0 101 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - -
Tnx very much for help again
Regards
Edoardo
regards klaus
regards klaus
Edoardo Serra wrote:
Hi guys, I'm having a problem with an OpenSER acting as registrar server and load balancer for many Asterisk servers. In a few words: "users are registering on openser and, when they want to make a call, OpenSER proxies the request to an Asterisk server with the dispatcher module" Here is the intended data flow (SIP goes through OpenSER and media goes directly to Asterisk) User <-- SIP --> OpenSER <-- SIP --> Asterisk User <-- RTP --> Asterisk Both, OpenSER and Asterisks have public IPs I already have a working setup of that and everything seems working correctly. I'm trying to replicate that setup on another site, same configurations of the boxes, same versions of OpenSER and Asterisk, etc... but I'm having monodirectional Audio. Having a look with tethereal I see that OpenSER, when the communication is answered, sends a SIP packet (200 OK) to the user indicating itself as media endpoint instead of the Asterisks. From that moment I see RTP packets flowing from the client to OpenSER This seems really strange to me because I just copied the same configurations file from a working setup to the new installation. Tnx in advance for help. Regards P.S.: Here is my openser.cfg ## $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ ## simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #children=4 #port=5060 fifo="/tmp/ser_fifo" #uid=nobody #gid=nobody # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/exec.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser") modparam("usrloc", "timer_interval", 120) modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/voip") modparam("uri_db", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "max_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("nathelper", "natping_interval", 20) # Ping interval 20 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("dispatcher", "force_dst", 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 > max_len ) { sl_send_reply("513", "Message too big"); exit; }; if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) { sl_send_reply("405", "Method Not Allowed"); exit; } if (!method=="REGISTER") { record_route(); }; if ((src_ip==xxx.xxx.xxx.xxx) || (src_ip==xxx.xxx.xxx.xxx)) { # IP of Asterisks if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; exit; }; if (nat_uac_test("3")) { if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) { fix_nated_contact(); force_rport(); setflag(6); # Mark as NATed } } # 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 (method=="REGISTER") { if (!proxy_authorize("domain", "openser_view")) { proxy_challenge("domain", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; };
if (method=="INVITE") { if (!proxy_authorize("domain", "openser_view")) { proxy_challenge("domain", "0"); exit; } if (!check_from()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } } # loose-route processing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); exit; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # ! Nathelper if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if ((src_ip!=xxx.xxx.xxx.xxx) && (src_ip!=xxx.xxx.xxx.xxx)) { # IP of Asterisks ds_select_dst("2", "0"); } if (!t_relay()) { sl_reply_error(); }; } # ! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }
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-- Klaus Darilion nic.at