sr-users November 2004

sr-users@lists.kamailio.org

203 participants
386 discussions

[Serusers] error with ser unstable from cvs

by Schweizer Laurent

Hello, I have downloaded ser from cvs (unstable) to test AVPops Module and I see that I have many errors (tested with eyepmedia and xlite). Nov 12 15:20:16 voip ser[24130]: ERROR: CSeq EoL expected Nov 12 15:20:16 voip ser[24130]: ERROR: parse_cseq: bad cseq Nov 12 15:20:16 voip ser[24130]: ERROR: get_hdr_field: bad cseq Nov 12 15:20:16 voip ser[24130]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_FORWARD header failed! Nov 12 15:20:16 voip ser[24130]: ERROR: CSeq EoL expected Nov 12 15:20:16 voip ser[24130]: ERROR: parse_cseq: bad cseq Nov 12 15:20:16 voip ser[24130]: ERROR: get_hdr_field: bad cseq Nov 12 15:20:16 voip ser[24130]: find_first_route(): Error while parsing headers Nov 12 15:20:16 voip ser[24130]: ERROR: CSeq EoL expected Nov 12 15:20:16 voip ser[24130]: ERROR: parse_cseq: bad cseq Nov 12 15:20:16 voip ser[24130]: ERROR: get_hdr_field: bad cseq Nov 12 15:20:16 voip ser[24130]: append_hf(): Error while parsing message Thanks Laurent

20 years

[Serusers] SIP and SER newbie

by S S

Hello everyone. I downloaded the latest version (0.8.14) of SER and installed it on my redhat linux server. I've not set up a sip server before so I'm new to this. 1.) Would you please point me in the direction to get started in configuring the SER server? I would like to test it out w/ an ip phone and windows messenger. I've been using some of the documentation that were available at the same ftp site. 2.) I noticed that it is preferable to install and use mysql. I already have mysql installed on my server, but the file "ser-mysql" could not be found in the latest directory on the ftp site. Where can I find that file to set up mysql for SER? 3.) What ports should I have open for SER to function if I'm behind firewall? 4.) Lastly, I would also like to set up the sms gateway module. How can I have access to that module? Thank you for your help. Sher

20 years

Re: [Serusers] SER + NAT (Allan, please?)

by LBE_SIP＠telefonica.net

> Ser doesn't send any media. You mean rtpporxy probably. Yes, you are right. Sorry for not being clear enough. > Some network dumps and your ser.cfg might help. ser.cfg is at the botton of the message. The scenario is as follows: The sip server is listening on 194.179.25.52. The user chano(a)wmserver.hi.inet on the machine sayuritra (private address: 192.168.1.34, public address: 81.35.36.166) calls to the user raquel(a)wmserver.hi.inet on hobbes (private address: 192.168.1.34, public addess: 80.32.97.245). The messages captured in the sip server are the following (irrelevant messages are omitted): 1) INVITE from sayuritra to server (81.35.36.166:20820 -> 194.179.25.52:5060): INVITE sip:raquel@wmserver.hi.inet SIP/2.0..Via: SIP/2.0/UDP 81.35.36.166:20818..From: "chano" <sip:chano@wmserver.hi.inet>;tag=a350c270-a7ff-4149-8e05-8af9f79e9e92..To: <sip:raquel@wmserver.hi.inet>..Call-ID: 5e5ba0ab-2321-4e87-b038-5a99fc37acdd@81.35.36.166..CSeq: 1 INVITE..Contact: <sip:81.35.36.166:20818>..User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Content-Length: 531....v=0..o=sayuritra 0 0 IN IP4 81.35.36.166..s=session..c=IN IP4 81.35.36.166..b=CT:1000..t=0 0..m=audio 20857 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..m=video 61674 RTP/AVP 34 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000.. (so sayuritra is waiting for incoming video from server on port 61674 and audio on 20857) 2) INVITE from server to hobbes (194.179.25.52:5060 -> 80.32.97.245:12595): INVITE sip:80.32.97.245:12595 SIP/2.0..Max-Forwards: 10..Record- Route: <sip:194.179.25.52;ftag=a350c270-a7ff-4149-8e05- 8af9f79e9e92;lr=on>..Via: SIP/2.0/UDP 194.179.25.52;branch=z9hG4bKdbfe.e72b2f84.0..Via: SIP/2.0/UDP 81.35.36.166:20818..From: "chano" <sip:chano@wmserver.hi.inet>;tag=a350c270-a7ff-4149-8e05-8af9f79e9e92..To: <sip:raquel@wmserver.hi.inet>..Call-ID: 5e5ba0ab-2321-4e87-b038- 5a99fc37acdd@81.35.36.166..CSeq: 1 INVITE..Contact: <sip:81.35.36.166:20818>..User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Content-Length: 550..P-hint: usrloc applied....v=0..o=sayuritra 0 0 IN IP4 81.35.36.166..s=session..c=IN IP4 194.179.25.52..b=CT:1000..t=0 0..m=audio 35106 RTP/AVP 9 7 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..m=video 61674 RTP/AVP 34 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000..a=n ortpproxy:yes.. (so server is waiting for incoming video from hobbes on port 35106 and audio on 20857) 3) 200 OK from hobbes to server (80.32.97.245:12595 -> 194.179.25.52:5060): SIP/2.0 200 OK..Via: SIP/2.0/UDP 194.179.25.52;branch=z9hG4bKdbfe.e72b2f84.0..Via: SIP/2.0/UDP 81.35.36.166:20818..From: "chano" <sip:chano@wmserver.hi.inet>;tag=a350c270-a7ff-4149-8e05-8af9f79e9e92..To: <sip:raquel@wmserver.hi.inet>;tag=54e09747-9530-456b-b0bf-e98bc4c72d3c..Call-ID: 5e5ba0ab-2321-4e87-b038-5a99fc37acdd@81.35.36.166..CSeq: 1 INVITE..Record-Route: <sip:194.179.25.52;ftag=a350c270-a7ff-4149-8e05-8af9f79e9e92;lr=on>..Contact: <sip:192.168.1.34:9843>..User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Content-Length: 528....v=0..o=hobbes 0 0 IN IP4 192.168.1.34..s=session..c=IN IP4 192.168.1.34..b=CT:1000..t=0 0..m=audio 47976 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..m=video 25608 RTP/AVP 34 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000.. (so hobbes is waiting for incoming video from server on port 25608 and audio on 47976) 4) 200 OK from server to sayuritra (194.179.25.52:5060 -> 81.35.36.166:20818): SIP/2.0 200 OK..Via: SIP/2.0/UDP 81.35.36.166:20818..From: "chano" <sip:chano@wmserver.hi.inet>;tag=a350c270-a7ff-4149-8e05-8af9f79e9e92..T o: <sip:raquel@wmserver.hi.inet>;tag=54e09747-9530-456b-b0bf-e98bc4c72d3c..Call-ID: 5e5ba0ab-2321-4e87-b038-5a99fc37acdd@81.35.36.166..CSeq : 1 INVITE..Record-Route: <sip:194.179.25.52;ftag=a350c270-a7ff-4149-8e05-8af9f79e9e92;lr=on>..Contact: <sip:80.32.97.245:12595>..User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Content-Length: 547....v=0..o=hobbes 0 0 IN IP4 192.168.1.34..s=session..c=IN IP4 194.179.25.52..b=CT:1000..t=0 0..m=audio 35108 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..m=video 25608 RTP/AVP 34 31..a=rtpmap:34 H263/90000..a=rtpmap:31 H261/90000..a=nortpproxy:yes.. (so server is waiting for incoming video from sayuritra on port 25608 and audio on 35108) After the connection has stablished, if we put a sniffer in sayuritra, we can see that there are ICMP 'destination unreachable' packets directed from server to sayuritra on port 25608, that is, the RTP proxy is directing the video packets to a closed port on sayuritra. Oddly, that port is the one where hobbes is listening on, that is, it seems that the RTP proxy is sending packets to the correct port in the wrong machine or, the other way other round, to the right machine in the correct port. Any ideas?. Thank you very much, Luciano Bajo Ser.cfg: # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ # machine has two network interfaces, listen only on the public one: listen=194.179.25.52 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 ---------------------------------- # Uncomment this if you want to use SQL database #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" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- 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) # -- 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper","rtpproxy_sock","/var/run/rtpproxy.sock") # ------------------------- 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 >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # 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(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request #append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request #append_hf("P-hint: outbound\r\n"); route(1); 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") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { #ppend_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # 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 (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # 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(); }; }

20 years

RE: [Serusers] "200 - Cancelling" Message Problem

by Ricardo Martinez

Hello Andrei. Thanks for your reply. I'm using Mediaproxy with the mediaproxy module from SER. I'm use client_nat_test to modify the rport. if (client_nat_test("1")) { log(1, "NAT: Requerimiento de IP privada --> fixed contact (en rutina principal)\n"); setflag(5); force_rport(); fix_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); }; Here, i'm attaching the complete debug from a call with a CANCEL request. (cancel_problem.txt) Can this problem be solved? or is more related with the endpoint? Thanks in advance. Best Regards Ricardo Martinez Ogalde -----Mensaje original----- De: Andrei Pelinescu-Onciul [mailto:pelinescu-onciul@fokus.fraunhofer.de] Enviado el: Viernes, 12 de Noviembre de 2004 5:25 Para: Ricardo Martinez CC: 'serusers(a)lists.iptel.org' Asunto: Re: [Serusers] "200 - Cancelling" Message Problem On Nov 11, 2004 at 12:52, Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > Hello List. > I'm still stuck with this problem. I really don't know if it is a > configuration problem or maybe a problem with SER, i was hopping that > someone could help me. I have a NAT'd client, when this client make a call, > and before the called party answer the call the call is canceled, I see a > "200 - Cancelling" message coming from SER to my NAT'd client. All the > previus messages goes to the correct port informed in the rport, but this > 200-Cancelling message goes to port 5060, so the message didn't reach the > endpooint. All the replies to a request are supposed to go to the rport. CANCEL is a different request. CANCEL will go to its request uri , in your case sip:2408196@sipproxy.mydomain.com. >It this a problem with the configuration or a problem with SER?. > Here is SIp debug for the final CANCEL message. A nat-config problem probably. Do you use nathelper? Do you use fix_nated_contact() for REGISTERs comming from natted UAs? Andrei

20 years

[Serusers] [Newbie question] - NAT/port messed up

by Thomas

Hello! I'm trying to make my 1st SIP-based system working. I use SER-0.8.14 on Linux Sarge. I have usually UAs behind NAT firewalls. I found a case, when UAs are working very oddly, I can't call from one of them, after sime time I can't reach one of them from outside, after some time when I call one UA, the 2nd is answering (and the same time when I call 2nd UA, the 2nd is answering). I have 2 UAs [Grandstream BT100 & X-Lite v2.0] behind a linux NAT firewall. Both UAs use the same STUN and SER server. I did some tcpdumps and I found an interesting thing: Internet Protocol, Src Addr: <FWPUBLICIP> (<FWPUBLICIP>), Dst Addr: <SERIP> (<SERIP>) User Datagram Protocol, Src Port: 1024 (1024), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: REGISTER sip:<MYDOMAIN> SIP/2.0 Method: REGISTER Resent Packet: True Suspected resend of frame: 31 Message Header Via: SIP/2.0/UDP <FWPUBLICIP>;branch=z9hG4bK0dbeeb499dbd416e From: "M70" <sip:70@<MYDOMAIN>;user=phone>;tag=d4de3110a63ddb36 SIP Display info: "M70" SIP from address: sip:70@<MYDOMAIN> SIP tag: d4de3110a63ddb36 To: <sip:70@<MYDOMAIN>;user=phone> SIP to address: sip:70@<MYDOMAIN> Contact: * Call-ID: 6ead5eb5744d407d@<IP PHONE LAN IP> CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream BT100 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Internet Protocol, Src Addr: <SERIP> (<SERIP>), Dst Addr: <FWPUBLICIP>77.150.18 (<FWPUBLICIP>) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP <FWPUBLICIP>;branch=z9hG4bK0dbeeb499dbd416e From: "M70" <sip:70@<MYDOMAIN>;user=phone>;tag=d4de3110a63ddb36 SIP Display info: "M70" SIP from address: sip:70@<MYDOMAIN> SIP tag: d4de3110a63ddb36 To: <sip:70@<MYDOMAIN>;user=phone>;tag=9ed85684aa54afbffef0a2d40b9935f4.71f6 SIP to address: sip:70@<MYDOMAIN> SIP tag: 9ed85684aa54afbffef0a2d40b9935f4.71f6 Call-ID: 6ead5eb5744d407d@<IP PHONE IP> CSeq: 100 REGISTER Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 <SERIP>:5060 "Noisy feedback tells: pid=25451 req_src_ip=<FWPUBLICIP> req_src_port=1024 in_uri=sip:<MYDOMAIN> out_uri=sip:<MYDOMAIN> via_cnt==1" What is interesting is that REGISTER message went out from port 1024 but 200 OK came to port 5060. Is this OK? I think here is some problem and this can make the phones working oddly. SER is configured with nathelper+rtpproxy. ... if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; ... It is basically the sample nathelper configuration. Could anybody give me a hint? If it is a well-known situation, please point me to some other resources, I'm really a new-comer in SIP/SER. Thank you in advance! Kind regard, Tamas J.

20 years

[Serusers] [mediaproxy] Memory usage

by Ezequiel Colombo

Hi, i have running mediaproxy 1.0 with Python 2.3 (compiled from source) in a Linux RedHat 8.0 kernel 2.4.18-14. After a few weeks running i see high memory utilization from mediaproxy.py process. I am loggin Vm usage to see if exist a leak. * Captured from /proc/PID/Status every 15 minutes. * The increment is done after each call processed. * I try to keep the process running several days to see if python free mem but never occur, always increment. ... VmSize: 7964 kB VmSize: 8780 kB VmSize: 8780 kB VmSize: 10104 kB VmSize: 11512 kB VmSize: 11792 kB VmSize: 11804 kB VmSize: 11820 kB VmSize: 11824 kB VmSize: 11828 kB VmSize: 12340 kB VmSize: 13580 kB VmSize: 14216 kB VmSize: 14216 kB VmSize: 14216 kB VmSize: 14216 kB VmSize: 14728 kB VmSize: 14812 kB VmSize: 14828 kB VmSize: 14852 kB VmSize: 14856 kB VmSize: 14856 kB VmSize: 15508 kB VmSize: 16036 kB ... Thanks Ezequiel

20 years

[Serusers] "200 - Cancelling" Message Problem

by Ricardo Martinez

Hello List. I'm still stuck with this problem. I really don't know if it is a configuration problem or maybe a problem with SER, i was hopping that someone could help me. I have a NAT'd client, when this client make a call, and before the called party answer the call the call is canceled, I see a "200 - Cancelling" message coming from SER to my NAT'd client. All the previus messages goes to the correct port informed in the rport, but this 200-Cancelling message goes to port 5060, so the message didn't reach the endpooint. It this a problem with the configuration or a problem with SER?. Here is SIp debug for the final CANCEL message. U xx.xx.148.244:64435 -> xx.xx.154.35:5060 CANCEL sip:2408196@sipproxy.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK2542303ca430 From: <sip:5555848110@sipproxy.mydomain.com>;tag=2542303ca4 To: <sip:2408196@sipproxy.mydomain.com> Call-ID: 252eb142-ae30-30f0-803c-0002a400f1e9(a)192.168.0.191 CSeq: 30 CANCEL Date: Thu, 16 Jun 2005 07:45:57 GMT User-Agent: AddPac SIP Gateway Content-Length: 0 Max-Forwards: 70 # U xx.xx.154.35:5060 -> xx.xx.148.231:5060 CANCEL sip:7072408196@xx.xx.148.231:5060 SIP/2.0 Record-Route: <sip:2408196@xx.xx.154.35;ftag=2542303ca4;lr=on> Via: SIP/2.0/UDP xx.xx.154.35;branch=z9hG4bKe6c7.e2501f15.0 Via: SIP/2.0/UDP 192.168.0.191:5060;received=xx.xx.148.244;branch=z9hG4bK2542303ca430 From: <sip:5555848110@sipproxy.mydomain.com>;tag=2542303ca4 To: <sip:2408196@sipproxy.mydomain.com> Call-ID: 252eb142-ae30-30f0-803c-0002a400f1e9(a)192.168.0.191 CSeq: 30 CANCEL Date: Thu, 16 Jun 2005 07:45:57 GMT User-Agent: AddPac SIP Gateway Content-Length: 0 Max-Forwards: 69 P-hint: H323-Side # U xx.xx.154.35:5060 -> xx.xx.148.244:5060 SIP/2.0 200 cancelling Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bK2542303ca430;received=xx.xx.148.244 From: <sip:5555848110@sipproxy.mydomain.com>;tag=2542303ca4 To: <sip:2408196@sipproxy.mydomain.com>;tag=8acf748d21b6bc66e44e951325ba3f1c-b28 c Call-ID: 252eb142-ae30-30f0-803c-0002a400f1e9(a)192.168.0.191 CSeq: 30 CANCEL Server: Sip EXpress router (0.8.14-3 (i386/linux)) Content-Length: 0 Warning: 392 xx.xx.154.35:5060 "Noisy feedback tells: pid=23757 req_src_ip=xx.xx.148.244 req_src_port=64435 in_uri=sip:2408196@sipproxy.mydomain.com out_uri=sip:7072408196@xx.xx.148.231:5060 via_cnt==1" This cause a endless CANCEL message flow coming from my endpoint. Hope that someone can help me here Thanks in advance Best Regards Ricardo Martinez

20 years

[Serusers] SER + NAT (Allan, please?)

by LBE_SIP＠telefonica.net

Hello, We are trying to get SIP Express Router and RTP proxy working for NAT traversal (as many of you, I think). We have tried all kind of configuration files, versions, etc. but the best result we get is always the same: the session is stablished, but the media are not. We have followed, step by step, Allan's help (thank you, I am the guy that asked for your ser.cfg) in his post on Nov 4th (http://lists.iptel.org/pipermail/serusers/2004-November/012737.html), but we are not able to get it working. The problem seems to be that SER does not send the media to the ports indicated by SDP, instead it sends them to wrong ports resulting in ICMP 'destination unreacheable' errors. As clients we are using two Windows Messenger 4.7 in two different ADSL (NATed) lines connecting to SER in a public Internet address. The linux box for SER is Red Hat 9.0 and file versions are the same as indicated in Allan's post. Any suggestions, ideas, ..., please? The deadline is really near and we do not know what else to do. Thanks in advance, Luciano Bajo

20 years

[Serusers] stateless/statefull processing and INVITE method retransmissions

by Michael Kaufmann

Hi, Ser is pretty new to me (about 3 weeks of reading about it and working with it), so naturally I have some questions: (attached - the configuration file I use) I saw many examples of configuration files were both SL and TM modules are loaded. In such cases is SL loaded only for REGISTER requests? How can I control which of SER actions are handled with TM and which are handled with SL? Is there a way of hooking SER answers to requests without TM? As I understand from this mail archive - INVITE is a special case in SER and even though the client doesn't retramit such requests, SER does. Is there a way to avoid this retransmission? Is there a way to control the interval time between one retransmission to another? It seems that sometimes these retransmissions occure even after the ACK is returned... Another starnge this is that my clients are on the same domain as the one SER is on and still record_route() seems to add many non usefull header lines for such INVITE messages. Is it a problem with my configuration file? Thanks in advance, Micky _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.com/

20 years

[Serusers] SER NAT problem

by VoIP

Hi, all I use sjphone behind NAT which can register with sip.iptel.org successfully. But fail to register with my own SER in public internet. I already setup nathelper and rtpproxy, and both rtpproxy and SER running properly. I don't know why register still fail. My ser.cfg is as follows, # # $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) 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 ---------------------------------- # Uncomment this if you want to use SQL database 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" # 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" # NAT support loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- 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) # -- 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") # -- rr params - # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # NAT support modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 10) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") #rtp proxy socket # ------------------------- 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 > max_len ) { 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; #Added by YangTao # attempt hand off to PSTN if (uri=~"^sip:[0-9]*@127.0.0.1") { prefix("5589712"); forward (202.157.148.229,5060); }; #End by YangTao # 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=~"myown.com") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myown.com", "subscriber")) { www_challenge("myown.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # break; #}; }; # 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(); }; } }

20 years

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sr-users November 2004