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37517 discussions

[Serusers] Multiple user registrations using same port number
by Alan 03 Nov '06

03 Nov '06

I am currently running load testing on SER and have a question regarding multiple contacts registered with the same IP and port number but different contact name. Will this work? Does each contact ID have to be defined with a different port number to establish a session? For example: Contact : sip:loadtest0100@10.180.0.71:10140 Contact : sip:loadtest0101@10.180.0.71:10140 Contact : sip:loadtest0102@10.180.0.71:10140 Contact : sip:loadtest0103@10.180.0.71:10140

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[Users] About forking and unregister
by Bill Zhang 03 Nov '06

03 Nov '06

Hi, Anyone know how openser works with following: 1. Forking. Often times, I may have one account register with multiple SIP client, does it support that? I don't seem to make it working. 2. Unregister and register. Nowadays, there are a lot of SIP client will send an Unregister message to the SIP server and after received response, it will send the new registration again, this is because when a client behind the same router reboot multiple times, it might have registration in the SIP server and only the latest one reflect the current registration and all previous registration may not be valid as the router may already close the mapping port, so in RFC3261, it defines client can send a registration with contact head contains "*" and expeiration timer is 0 to unregister the previous registration. It does not seem that OPENSER can handle it. I know SER handles it. 3. Can I specify OPENSER to only send the INVITE (or other SIP request) to the latest AOR for a particular account? Thanks in advance. Bill

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[Users] mediaproxy pulse / jitter and dropout
by Mark Price 03 Nov '06

03 Nov '06

I installed the openser.cfg at http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy (modified to authenticate against my domain, and to use exit instead of break). Calls between two phones on the domain work, but there is a lot of jitter. Well, not exacty jitter, but kind of a pulsing sound in the background of the call. Also, will occasionally drop out, and then come back after a little bit.

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[Serusers] how to search in usrloc?
by sandrinho1976＠libero.it 03 Nov '06

03 Nov '06

Hi everyone. I'm facing this problem when configuring a Ser as a proxy and a registrar. I have the Ser and a IP pbx: the pbx registers itself on the proxy as pbx@<pbx_ip> (Contact header). I had to manage INVITEs messages for incoming call that are going towards the Ser, sent by UACs in the network. INVITEs are coming in the form <telephone_number>@<ser_ip> (Req-URI): Ser had to decide if <telephone_number> is managed by the IP pbx (it has this information) and eventually, do a rewrite uri to <telephone_number>@<pbx_ip> and then forward the message to pbx. So far, it's all ok. Unfortunately, i have to handle also this situation: the ip pbx is down. At the present, the INVITE is forwarded towards the pbx, the ip pbx does not respond and after 10 seconds the requesting UACs sends a CANCEL. This latency is not acceptable. I'm looking for a way to implement this: after the rewrite uri to <telephone_number>@<pbx_ip>, look if I have a contact in usrloc with <pbx_ip> (i.e. the pbx is still registered): if not, send immediately a response 4xx to UAC. The problem is that i'm not able to access the usrloc db from the routing script... any hint? Thanks a lot...

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[Users] two invites, different session description => broken call
by Mark Price 02 Nov '06

02 Nov '06

I have openser configured with asterisk as following: If a sip REGISTER packet arrives, fix_nated_register() save("location") exit; If a sip packet arrives at openser with a source ip != asterisk ip, t_relay(to asterisk) If a sip packet arrives from asterisk, lookup("location"); force_rport() fix_nated_contact() t_relay(); The asterisk dial plan says the following: if inbound traffic arrives from openser's ip address, Dial( SIP/dstuser@dstdomain). phones a and b are on separate networks behind a firewall, openser and asterisk are on public ip addresses. The sip traffc seems to work just fine. I.e. all the handshakes seem to be happening as they should. However, rtp traffic does not. Whether audio traffic will transmit or not is a crap shoot. Sometimes, if a calls b, b hears a but not vice versa, while if b calls a, audio is two way. In this case, one audio stream is going through asterisk, the other is being directed to go point to point. Sometimes, a calls b and b hears a, and a hears b for a second but a second INVITE comes to phone B that causes it to redirect rtp to be point to point. Sometimes there is no audio. Sometimes, everything works fine. At one point, rtp from a was going to asterisk, but asterisk was not sending the rtp on to b, and b was trying to send traffic point to point.

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Re: [Users] TLS: what is encrypted in TLS?
by Ferianto siregar 02 Nov '06

02 Nov '06

Dear Witt, Thank you very much for your help before. Witt, may I ask you again,please? I have downloaded ssldump.tar.gz packet and installed it. As the INSTALL note in ssldump, I have installed libcap.tar.gz (from www.tcpdump.org) and openssl. I have installed it by using commands: # tar -zxvf ssldump-0.9b3.tar.gz # cd ssldump-0.9b3 # ./configure --with-openssl #make #make install All installation process can run sucessfully (nothing error message shown). But, why when I try to run ssldump, I got nothing. There is any mesage shown when I tried to run #ssldump -i eth0 What happens? Did I do wrong installation process? or I have missed something? I do hope you can help me.Please help me.. Thank you very much for your help Regards, Ferianto Note: Actually, when I run ./configure --with-pcap, I got error message. The error message said that there is no pcap file. Why? Is lipcap packet that I got from www.tcpdump.org (as I read from ssldump INSTALL), same with pcap file ? or I have to download pcap ? Steffen Witt <witt.steffen(a)googlemail.com> wrote: Hello Ferianto, TLS is used for encrypting your SIP signaling, not the media data. You should install "ssldump" on your server to see each step of the TLS handshake. Maybe you have a closer look to this short SSL tutorial: http://www.eventhelix.com/RealtimeMantra/Networking/SSL.pdf Best regards Steffen 2006/10/30, Ferianto siregar : > Dear all, > > Thank you for this chance. > All, I have tried to learn about how can TLS can secure communication in > VoIP. > I must to understand it in order to responsible my openser server (working) > with TLS to my lecturer. > But, I have still doubt. > Actually, what is encrypted in TLS system so the communication become > secure? Does it encrypt the data (voise packet)? or The traffic (so the > sniffer can not get the data? OR anything else? > > I do hope anybody can help me :) > Thanks, > > > Ferianto > > > Note: I cryptography tutorials, I read that the cryptography system secure > the message (text) by encrypting it using public key and private key. So, > How about TLS in VoIP? > > > ________________________________ > Everyone is raving about the all-new Yahoo! Mail. > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > --------------------------------- Get your email and see which of your friends are online - Right on the new Yahoo.com

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[Users] sarge didnt install avp_radius
by Zoran Milic 02 Nov '06

02 Nov '06

Hi, I changed my linux distro to debian (from slack) so i can install openser from packages, but aptitude failed to install avp_radius. When I install from source, avp_radius.so is created, after tweaking the Makefile. What is the problem, and how do i get avp_radius module when i install from aptitude/apt-get/dpkg ? best regards, Zoran

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[Users] determining originating address
by Mark Price 02 Nov '06

02 Nov '06

What is the best way to detect whether an INVITE message came from a certain IP address? I.e. if a call comes from a certain asterisk box? I have thought of using the textops search(), but this seems a little clumsy. Perhaps there is a way to get directly at the via headers? Or is there a keyword of corresponding functionality to the uri keyword? Thanks, Mark Price

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[Serusers] please help me out
by vijay tiwari 02 Nov '06

02 Nov '06

hi all i have problem in configurations of mediaproxy module i have download mediaproxy in my ser server and i have make one file name(mediaproxy.ini) in mediaproxy dir and do some change and run this command mediaproxy start nothing can heppand please help me out on this because at this time i am in very bad conditions thanks vijay --------------------------------- Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Everyone is raving about the all-new Yahoo! Mail.

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[Serusers] No Media with Nated Client and rtp proxy
by Ryan Churches 02 Nov '06

02 Nov '06

I just posted about a problem with rtpproxy, but i was able to correct that. now ser is able to connect to rtpproxy, but media is still being lost. someone suggested i put my external ip in the listen= value at the top, but that breaks remote registration. I am following the rtpproxy section of the "Getting Started with Ser Docs" http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html so one would expect i would have ser.cfg set up correctly, but obviously something is wrong. here is some ngrep output. 10.0.0.10 is SER. 10.0.0.110 is a client LOCAL to ser, and 10.0.0.100/67.84.215.54 is a remote client behind yet another nat U 10.0.0.10:6060 -> 10.0.0.110:9267 INVITE sip:1000@10.0.0.110:9267 SIP/2.0. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. To: <sip:1000@bmy.gotdns.org>. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Contact: <sip:1001@67.84.215.54:9170>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3004w stamp 16863. Content-Length: 285. . v=0. o=- 105355536 105355589 IN IP4 10.0.0.100. s=eyeBeam. c=IN IP4 10.0.0.10. t=0 0. m=audio 20026 RTP/AVP 100 6 0 8 3 18 5 101. a=alt:1 1 : 7D3DDAD1 578F25CA 10.0.0.100 9172. a=fmtp:101 0-15. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=nortpproxy:yes. # U 10.0.0.110:9267 -> 10.0.0.10:6060 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.10:6060 -> 67.84.215.54:9170 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.110:9267 -> 10.0.0.10:6060 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.10:6060 -> 67.84.215.54:9170 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. and here is my ser.cfg which ive also attached. # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=10.0.0.10 alias=hostname.dyndns.org port=6060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:pw@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #From Getting Started modparam("auth_db|uri_db|usrloc", "db_url", "mysql://user:pw@localhost/ser") # ----------------- setting module-specific parameters --------------- # -- 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("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) # -- 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; }else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; #======= call type processing section if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" || isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #------------------------- #REGISTER Message Handler #------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #---------- #--INVITE Message Handler #----------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { #NAT Traversal if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Contact-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

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