sr-users February 2006

sr-users@lists.kamailio.org

189 participants
374 discussions

[Serusers] SER RTP QoS Monitoring
by Aisling O'Driscoll 02 Feb '06

02 Feb '06

Hi, I want to be able to extract QoS parameters such as jitter, delay and packet loss from VoIP calls made using my SER(both peer to peer for my public clients and also for clients where the voice will be routed with mediaproxy). I would like to develop something as opposed to purchasing a software package. >From searching the Internet the last few days I have seen that the following options seem to be available: 1) RTCP RTCP stream will contain this information. The problem is that sniffers cannot clearly distinguish RTCP packets from other UDP packets unless something like an IP address or specific UDP port is supplied. I had considered using TCPDUMP but then where do I position this in the network when voice is going peer to peer between client and I'm not sure how this can work to monitor all calls when I might not be aware of the end user IP addresses. 2) SIP INFO extension This also seem to be an option but most off the shelf phones don't seem to support this and this would required modification of a SIP ua. 3) SNMP I thought maybe SNMP might be an option but the SER snmp module no longer exists... Does anyone have any comments on the above? Are the statements that I made correct or can anyone think of other ways to monitor the voice QoS? I am trying to understand how commercial applications have accomplished this. Many thanks in advance for advice. Aisling. >---- Original Message ---- >From: clona(a)cyberhouse.no >To: ashling.odriscoll(a)cit.ie >Subject: Re: [Serusers] Status of SER SNMP module >Date: Sun, 29 Jan 2006 20:30:02 +0100 > >> >>Hi Aisling, >> >>The SNMP module is dead as a killed turtle :( >> >>-Atle >>* Aisling O'Driscoll <ashling.odriscoll(a)cit.ie> [060129 20:26]: >>> Hello, >>> >>> I'm just wondering what the current status of the SER SNMP module >is? >>> Is it currently supported and is there somewhere I can find >>> documentation on it? >>> >>> Many thanks, >>> Aisling. >>> >>> >>> >>> -------------------Legal >Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and >intended only for the person to whom it is addressed. Its contents >may be protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)iptel.org >>> http://mail.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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[Serusers] high-availibility setup using f5 bigip
by Jack Wei 02 Feb '06

02 Feb '06

hi, I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do load-balancing. I'm using Asterisk as a voicemail application only and have successfully integrated SER with Asterisk without the switch. But when I try to use the switch as a load-balancer, I get lots of NAT problems. Does anyone know how to setup the switch and SER/Asterisk properly? Thanks, Jack __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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RE: [Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Then where do i have to insert the codes in ser.cfg, the timeout for forward to voicemail? I tried some places but not successfull My ser.cfg is from onsip ser mediaproxy examle; Here, # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" 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/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") 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/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; 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] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; 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 (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; 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 Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } 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(); }; } Toygun >Read the more recent SEMS documentation. vm.so was deprecated. >You now use other methods (depending on whether you're using a fifo or a unix socket in SEMS). >For instance, I use: >if(!t_write_unix("/tmp/am_sock","voicemail")) { > t_reply("500","error contacting sems"); >}; On Wed, 1 Feb 2006 20:44:36 +0200, Toygun Mavinil wrote > Hi, > I installed ser 0.9.6 on fc4 > Compiled and installed with all modules but vm.so module is missing. > Looked at the source modules directory, there is no vm directory or file. > I installed sems from cvs > Sems is working > > Also > Do i have to add down lines to ser.cfg > > loadmodule "/home/srouter/sip_router/modules/vm/vm.so" > modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("vm", "email_column", "email") > modparam("vm", "subscriber_table", "subscriber") > modparam("vm", "user_column", "username") > modparam("vm", "domain_column", "domain") > > if(method=="ACK" || method=="INVITE" || method=="BYE"){ > > if (!t_newtran()) { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new transaction"); > break; > }; > > t_reply("100","Trying -- just wait a minute !"); > if(method=="INVITE"){ > log("**************** vm start - begin ******************\n"); > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm start - end ******************\n"); > } else if(method=="BYE"){ > log("**************** vm end - begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm end - end ******************\n"); > }; > break; > }; > if (method=="CANCEL") { > sl_send_reply("200", "cancels are junked here"); > break; > }; > sl_send_reply("501", "method not understood here"); > > > > > > > Toygun >

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FW: [Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

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[Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Hi, I installed ser 0.9.6 on fc4 Compiled and installed with all modules but vm.so module is missing. Looked at the source modules directory, there is no vm directory or file. I installed sems from cvs Sems is working Also Do i have to add down lines to ser.cfg loadmodule "/home/srouter/sip_router/modules/vm/vm.so" modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("vm", "email_column", "email") modparam("vm", "subscriber_table", "subscriber") modparam("vm", "user_column", "username") modparam("vm", "domain_column", "domain") if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); Toygun

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[Users] Forward requests based on userid
by Mouncif Benziane 01 Feb '06

01 Feb '06

Hi folks, I wanted to route sip requests based on user id with it's appropriate Gateway IP address which is stored in Mysql database , is this possible? Database structure: userid,gateway 617978XXX 64.80.XXX.XXX 78189XXXX 64.100.XXX.XXXX example: if someone call 617978XXX he will be routed to it's approriate gateway at present I am using static route like: if (uri=~"^sip:617978XXX@") { rewritehost("64.80.XXX.XXX"); } if (uri=~"^sip:78189XXXX@") { rewritehost("664.100.XXX.XXXX"); } etc,.... I am afraid the config file gonna get larger if I add more than 500 users(DIDs), I am looking for an alternative to store DID numbers in database (Mysql or text) and route them accordingly with the appropriate Gateway address. Thank you so much and I will appreciate if you can help me out! Ben __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] check if transaction exists
by Andreas M. 01 Feb '06

01 Feb '06

Hi all, i want to introduce following in my configuration: CallerA calls CallerB -> ringing CallerC calls CallerB -> reject during this transaction, without that CallerB picks up the phone, CallerC should not be able to call CallerB during this transaction, After this transactions, CallerC can call CallerB also. Any ideas how to check if there is an ongoing transaction ? regards, Andreas M.

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[Serusers] Re: 200 OK/SDP and ACK/SDP problem
by Carla Ares 01 Feb '06

01 Feb '06

> > Hi all, > I have a problem with audio, when the destination of a call is natted and > the other side is public. This problem is cause because the SDP info only is > present in the 200 OK and in the ACK message. > > I'm using ser (8.14) with mediaproxy (1.2.1). > > I have an H323 to SIP translator (and SIP to H323 too) connected to the > PSTN and to the SIP Proxy (SER). > H323 is the origin of the call (public client), and SIP UAC is the > destination (Natted client). > My proxy is working in state full mode and has a public IP. > > The H323 side is using slow start, so when it starts the transaction with > an INVITE, it will not send the SDP info (it will send it in the ACK > message). > The proxy forwards this message to the destination. > SIP UAC, starts ringing. When we hooks off, the SIP UAC send an 200 OK > with SDP message to the H323 side. > > This is the 200 OK / SDP that the proxy receives from the SIP side. > > No. Time Source Destination Protocol > Info > 488 28.488811 200.68.89.2 200.68.89.12 SIP/SDP > Status: 200 OK, with session description > Message body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): 700600 6015 6015 IN IP4 > 192.168.0.101 > Owner Username: 700600 > Session ID: 6015 > Session Version: 6015 > Owner Network Type: IN > Owner Address Type: IP4 > Owner Address: 192.168.0.101 > Session Name (s): AddPac Gateway SDP > Connection Information (c): IN IP4 192.168.0.101 > Connection Network Type: IN > Connection Address Type: IP4 > Connection Address: 192.168.0.101 > Time Description, active time (t): 0 0 > Session Start Time: 0 > Session Stop Time: 0 > Media Description, name and address (m): audio 23018 RTP/AVP > 18 8 0 101 > Media Type: audio > Media Port: 23018 > Media Proto: RTP/AVP > Media Format: ITU-T G.729 > Media Format: ITU-T G.711 PCMA > Media Format: ITU-T G.711 PCMU > Media Format: 101 > > Following the ser.cfg, this 200 OK / SDP will make the proxy to use > mediaproxy (use_media_proxy()), because the destination user is natted. So > the mediaproxy module, will generate a lookup command to the > proxydispatcher.py. > > proxydispatcher[30535]: command lookup 2281401749(a)200.68.89.10 > 192.168.0.101:23018:audio 200.68.89.2 200.68.89.10 remote 200.68.89.10unknown AddPac=20SIP=20Gateway info= > from:1150316660@200.68.89.10,to:1152464490@200.68.89.10 > ,fromtag:3512844671,totag:7b07717a4 > proxydispatcher[30535]: warning: trying to lookup session with > non-existent id: '2281401749(a)200.68.89.10' > > The proxydispatcher does not recognize that command because it has not > generate the session before. So, the proxy forwards the 200 OK /SDP message > without changing the SDP info. When the H323 sides receives that info, it > thinks that it has to send RTP to the private IP. > > Then the H323 sends the ACK with SDP. > > Then, we cannot ear audio in both sides. > > Here is the problem, because, I didn't receive an INVITE with SDP that can > create the session into the dispatcher. > > > A is public, B is natted > > side A Proxy side B > > INVITE without SDP > ----------> > 100 Trying > <---------- > > INVITE without SDP > ----------> > > 100 Trying > <---------- > 180 Ringing > <---------- > 180 Ringing > <---------- > > 200 OK SDP (B private IP) > <---------- > *1* > 200 OK SDP (B private IP) > <---------- > > ACK SDP (A public IP) > ----------> > *2* > ACK SDP (mediaproxy public IP) > ----------> > > ...................................................... > RTP from B > <---------------------- > *3* > ...................................................... > > > *1* The proxy must replace the private IP of the SDP, with the ip of the > mediaproxy. The looku prequest does not work, because no session was found > *2* the proxy generates the session in the mediaproxy. So B thinks that > the RTP must go to the mediaproxy. But A never knows the mediaproxy address. > *3* RTP from A goes to anywhere, because A does not know where it is the > private adrress of B. > > Do you know a solution to this? Is the sequence of the message all right? Regards. Carla

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[Serusers] Here I go again...Call type processing Section problem?
by Juan 01 Feb '06

01 Feb '06

Hi. As I had no answers to this question, here it is again. Till now I was happy using SER handling communications between UAs using nathelper and RPT proxy. Now I changed the ser.cfg script to the one with PSTN support and none of the UAs get registered. I am getting a 403 Forbidden. Analyzing the script I understood that this 403 was coming from the Call Type Processing Section so I changed if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { and so on to if (uri!=myself) and now the UAs are registering again. But of course I can't make calls. So I would like to know how this< is_uri_host_local and is_from _local> work and what should I do to make them allow my UAs to register. I am attaching the ser.cfg that doesn't work. Thanks in advance Juan Ferrari

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[Users] TM bug?
by Greg Fausak 01 Feb '06

01 Feb '06

I am running a proxy pushing 2 million calls a day. Every now and then, I drop 5 or 10 BYE messages. I've attached the sip_scenario log file, F16,F17,F20 all send BYE messages to 12.46.104.62. F21 is sending the BYE form 12.46.104.62 to 12.46.104.252. The OK comes back, but is never relayed. BYEs continue to retry until 12.46.104.62 finally gives up and sends a Request timeout 408 in frame F38. So, F38 is evidence that the transaction started. Right? Why are all of the BYEs ignored, and why isn't the OK relayed? I have xlog output from 12.46.104.58 indicating each time a BYE is loose-routed to 12.46.104.62. The syslog file shows each attempt (F15,F18,F19,F23, etc). I also have xlog output from 12.46.104.62, however, the syslog on that machine never shows receiving or loose-routing a BYE with this callid. It seems like receiving more than one BYE in a short period of time is screwing up the t_relay()??? The calls are being generated with sipp, and 999,990 times out of a million it works. ---greg -- Greg Fausak greg(a)thursday.com

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sr-users February 2006