sr-users January 2011

sr-users@lists.kamailio.org

66 participants
111 discussions

Kamailio Devel Training, Barcelona - update

by Daniel-Constantin Mierla

Hello, a quick update about the next training in Barcelona, Feb 10-11, 2011, focusing on Kamailio development. We got a bigger room (for second time :-) ), so we increased the capacity to 28 seats. We are already 23 people in the class and several more joining the dinner Thursday evening, so there is going to be a great time. If you plan to come, register asap, there is no time to get a bigger room, so once we are full booked this time, we close the participants list. You can see more details at: http://www.kamailio.org/w/2011/01/kamailio-development-training-barcelona-f… Also, do not forget Fosdem next weekend, it is a bunch of us going there, two talks about Kamailio, come and say hi: http://www.kamailio.org/w/2011/01/social-networking-event-brussels-belgium/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

13 years, 9 months

Kamailio as a routing engine

by Gang Liu

Hello, I wrote a H.323/SIP IWF program before, it was based on B2BUA framework. Currently I am planning to use Kamailio as internal routing server and let IWF become a class 4 soft switch.That is Inbound ---- H.323/ SIP -- > ---- SIP routing request-----> Kamailio IWF/SBC | Outbound <--- H.323/SIP -- <--- SIP ----------------------------- Is it ok to store all voip provider's information at database of Kamailio and pass all to IWF at private SIP headers and let IWF as a outbound proxy? Is Kamailio as a SIP redirect server or it is better to stay at call signaling path until call session ended? It is great to share any information about this. regards, Gang

13 years, 9 months

Load_gw failed

by Thiago Maluf

Hi Kamailio Users, I'm trying to use LCR module in Kamailio 3.1.1 and I', having some problems with this function (load_gws). I'm my kamailio.cfg , I have this route: # SEARCH DEST route[NUMBER] { xlog("L_ALERT", "NUMBER IS $rU \n"); if ($rU=~"^55211020[0-9]{4}$") { # IS MY USERS route(LDAP_NUM2USER); } else { # NOT IS MY USER if (!load_gws(1,$rU)) { xlog("L_ALERT", "LOAD LOCAL GW FAILED $rU \n"); route(DSER); return; } else { xlog("L_ALERT", "LOAD GW DEBUG $avp(i:709)\n"); } ; if (!next_gw()) { xlog("L_ALERT", "NEXT LOCAL GW FAILED\n"); sl_send_reply("503", "Service not available - No gateways"); return; }; setflag(29); route(RELAY); return; }; }; When my call arrive to load_gws, I have this problem: (2198) ERROR: <core> [action.c:1158]: runtime fixup failed for load_gws param 2 (2198) ERROR: <core> [action.c:1511]: run action error at: /etc/kamailio/kamailio.cfg:472 I already saw the Kamailio Code and this error doesn't make sense. Would somebody Help me with it? Very thanks, Thiago Maluf. -- ---------------------------------------------------------------- Thiago Maluf Resende Tel: +55 21 9700-9113 e-mail: malufrj(a)gmail.com

13 years, 9 months

SIP Recorder

by Danny Dias

Hello my friends, I have a requeriment, which indicates that i have to record every SIP conversation between peers (also for callings to the PSTN); the recording server will be built for our company following this requeriments (also requested for the client): My doubt is: How can i handle sip conversations recording when all the calls are passing through a Proxy Server? I do understand that the media is always peer to peer and the signaling goes through the Proxy, but in this case the media not only has to pass between the peers because it must be recorded. How should i handle this?

13 years, 9 months

Kamailio CFG file size ??

by Jijo

Hi All, is there any limitation on the size of kamailo.cfg file. When i tried to add few avp_subst, then the kamailio starts showing up the error out of package memory on receiving SIP message. If i remove those lines then everything works fine.. avp_subst is simple one. Our kamailio cfg file is quite big.. Please let me know how to debug this issue? Here is the error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_row.c:116]: no private memory left 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:293]: could not allocate row 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:258]: failed to convert row #1 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_res.c:75]: failed to convert rows 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: db_postgres [km_dbase.c:416]: error while converting result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [db_query.c:137]: error while storing result 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: avpops [avpops_db.c:333]: cannot do the query 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: maxfwd [mf_funcs.c:62]: parsing MAX_FORWARD header failed! 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: rr [loose.c:98]: failed to parse headers 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: pv [pv_core.c:1761]: do action failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:02-05:00 [err] sipserver: ERROR: <core> [msg_translator.c:2229]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: rr [loose.c:70]: failed to parse To header field 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: siputils [checks.c:64]: To parsing failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:831]: ERROR: do_action: memory allocation failure 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [action.c:1511]: run action error at: :0 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1740]: do action failed) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:170]: get_hdr_field: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1381]: error parsing headers 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [data_lump.c:318]: ERROR: del_lump: out of memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: textops [textops.c:1081]: no memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [route.c:1846]: ERROR: eval_elem: bad or missing From: header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [backLinkMode] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_shv.c:367]: no more pkg mem 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/msg_parser.c:342]: ERROR:parse_headers: memory allocation error 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:400]: cannot parse To header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [temp] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [lvalue.c:358]: setting pvar failed 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [node] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_svar.c:117]: out of pkg mem! 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:1640]: error - cannot set svar [binding] 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:916]: osb_set_var_char(): setting profile PV failed. value INVALID (char) 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: pv [pv_core.c:418]: cannot parse From header 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: osb [osb.c:1257]: osb_get_osv_current_binding(): no binding available 2011-01-26T15:18:07-05:00 [err] sipserver: ERROR: <script>: R1 - binding IP=0 not available

13 years, 9 months

NAT Problem in Asterisk Realtime + Kamailio config

by andrzej.ciupek＠asterisk.edu.pl

Hello I have started to use configuration like: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb I works fine, but I have problem with dealing with NAT. kamctl ul show gives me: AOR:: andrzej_104 Contact:: sip:andrzej_104@10.5.0.94:14368;rinstance=9c2b256a13210aa7 Q= Expires:: 46 Callid:: ZDUwYzIzMDNiZTI3MjkyNmFmOWEyOTI2ODcyNjVmNDg. Cseq:: 60 User-agent:: eyeBeam release 1102q stamp 51814 State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 5087 AOR:: andrzej_103 Contact:: sip:andrzej_103@10.5.0.165:5060 Q= Expires:: 926 Callid:: 11631436-5060-1(a)10.5.0.165 Cseq:: 2257 User-agent:: Grandstream HT-502 V1.2A 1.0.3.10 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:77.xx.yyy.zzz:5060 Methods:: 6879 So when I have incomming call from Asterisk to Kamailio, Kamailio start to send call to LAN IP. I have tried to use help from: http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… But I have never get: Received:: and Cflag:: 192 it is always Cflag:: 0. field in my kamctl ul show. In my configuration Asterisk is Callweaver and is on the other server. I use rtpproxy-1.2.1 and kamailio-3.0.1 and mofified config from link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Please help.

13 years, 9 months

RLS question

by "Andrés S. García Ruiz"

Hi, I'm trying to create a RLS "rls-services" document with a contacts-list inside it. Kamailio states that "No xcap document found for uri:x". When creating a subscribe I must set the uri to the service uri. What I want to know is the sql query Kamailio does in order to get the contacts list for that uri. For example: XCAP document: "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" + "<rls-services xmlns=\"urn:ietf:params:xml:ns:rls-services\"" + " xmlns:rl=\"urn:ietf:params:xml:ns:resource-lists\">" + "<service uri=\"sip:restricted_areas_presentities@open-ims.test\">" + "<list name=\"friends\">" + "<rl:entry uri=\"sip:testuser01@open-ims.test\"/>" + "<rl:entry uri=\"sip:testuser02@open-ims.test\"/>" + "</list>" + "<packages>" + "<package>presence</package>" + "</packages>" + "</service>" + "</rls-services>"; Is stored in /xcap-root/rls-services/users/sip:restricted_areas@open-ims.test/index The subscribe is from sip:restricted_areas@open-ims.test to sip:restricted_areas_presentities@open-ims.test What am I doing wrong? Thanks, Andrés.

13 years, 9 months

[kamailio-users] kamailio on amazon

by Chandrakant Solanki

Hi I have installed Kamailio with MediaProxy and asterisk on Amazon Server.. While kamailio/MediaProxy and Asterisk both running on different amazon's instance. Kamailio : 1.5.0-notls MediaProxy : 2.3.8 Asterisk : 1.6.2.6 Firewall port is opened for mediaproxy from 10000-20000 (UDP), 5060 (UDP) etc on both amazon machine.. while I tried to play an audio... it plays sound file but unable to found audio on device. when I put tcpdump on asterisk machine it gives following error... # tcpdump -i eth0 udp portrange 10000-20000 -w test1.pcap 01:39:52.850279 IP (tos 0xb8, ttl 64, id 59, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 3cfd!] UDP, length 32 01:39:52.870279 IP (tos 0xb8, ttl 64, id 60, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 97fe!] UDP, length 32 01:39:52.890281 IP (tos 0xb8, ttl 64, id 61, offset 0, flags [DF], proto: UDP (17), length: 60) ip-W.X.Y.Z.compute.internal.15246 > xyz.com.10010: [bad udp cksum 75c0!] UDP, length 32 Any Idea..!! -- Regards, Chandrakant Solanki

13 years, 9 months

Re: [SR-Users] fix_nated_sdp()

by Bernhard Suttner

Hi, thanks for your response. I think I understand the scenario very good - but im not so familiar with the fix_nated_sdp() functionality. Currently the SDP will be modified with the source IP of the message. This does work good but currently I only do that for some special devices (matching the User-Agent / Server string). I want to do that fix_nated_sdp() in all cases and not only for some special devices. The FreeSWITCH RTP-Auto Adjustment does work in nearly every case but in one (currently known) scenario it does not. Maybe I should just give it a try to do fix_nated_sdp() in combination with nat_uac_test (RFC1918 private address in SDP). Thanks a lot for this discussion. Best Regards, Bernhard ----- Original Message ----- From: Ovidiu Sas [mailto:osas@voipembedded.com] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: Carsten Bock [mailto:lists@bock.info], sr-users(a)lists.sip-router.org Sent: Tue, 25 Jan 2011 20:38:14 +0100 Subject: Re: [SR-Users] fix_nated_sdp() > You need to experiment with nathelper module and while doing that, you > need to inspect the SIP messages in order to understand how it works. > Only when you fully understand your scenarios, you will be able to > properly configure, debg and setup kamailio. > > Best thing to do here is to use rtpproxy and relay media if needed. > > Another option would be to let the SDP untouched and let freeswitch do > it's magic (if it can). Not sure if this will work in all scenarios. > > Calling fix_nated_sdp() on the SBC will definitely not work (you are > just messing with the IP on the SDP and you don't know if the real RTP > port on the NATed side will match the advertised port in your "fixed" > SDP). > > > Regards, > Ovidiu Sas > > On Tue, Jan 25, 2011 at 1:23 PM, Bernhard Suttner > <bernhard.suttner(a)winet.ch> wrote: > > Hi, > > > > thanks for your answer. > > > > The question was about, if the fix_nated_sdp() (in the example below) > could maybe break something. I am sure, that the fix_nated_sdp() would work > in "some" cases but I am not sure, if the function could break something: > > > > > > if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > > > > and within the onreply route: > > > > if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > > > > > > Or is there a better solution to fix these damend NAT scenarios? > > > > Best regards, > > Bernhard > > > > > > -----Ursprüngliche Nachricht----- > > Von: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Im > Auftrag von Carsten Bock > > Gesendet: Dienstag, 25. Januar 2011 18:56 > > An: Bernhard Suttner > > Cc: sr-users(a)lists.sip-router.org > > Betreff: Re: [SR-Users] fix_nated_sdp() > > > > Hi Bernhard, > > > > have you checked the SDP which is going to and from the > > Freeswitch-Server? Is it modified? > > > > If yes: You may want to check your NAT-Firewall. A working (but not > > nice) solution might be to relay the RTP through an proxy in this > > case... > > If No: Verify the result of "nat_uac_test": Why is it returning "false"? > > > > Carsten > > > > 2011/1/25 Bernhard Suttner <bernhard.suttner(a)winet.ch>: > >> Hi, > >> > >> someone has an idea? > >> > >> Thanks in advance! > >> > >> Best regards, > >> Bernhard > >> > >> -----Ursprüngliche Nachricht----- > >> Von: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Bernhard > Suttner > >> Gesendet: Montag, 24. Januar 2011 20:38 > >> An: sr-users(a)lists.sip-router.org > >> Betreff: [SR-Users] fix_nated_sdp() > >> > >> Hi, > >> > >> I have a question about the fix_nated_sdp() function. Different devices > (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() > the message to a pool of FreeSWITCH Servers. > >> > >> The scenario does work quite well but: > >> > >> if a call is initiated from one of the freeswitch server, pass it through > kamailio to the PBX and the PBX does have a forwarding back to kamailio and > in the end to freeswitch I have no audio because the PBX is behind a NAT and > uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. > Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check > if IP in incoming packet is different to the IP in the SDP) to detect the > real IP address. > >> > >> Therefore I have to use fix_nated_sdp(). I am not sure, what this > function could break therefore I want to be very sure and use it only, if > really necessary. Test which User-Agent or Server the peer (in this example > the PBX) has, is not really possible because some devices does not send this > header. > >> > >> What do you think about the follwoing pseudo code: > >> > >> if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) > >> > >> and within the onreply route: > >> > >> if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) > fix_nated_sdp(3) > >> > >> Somebody has a better idea to handle this? What could go wrong? > >> > >> Thanks in advance. > >> > >> Best regards, > >> Bernhard Suttner > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users(a)lists.sip-router.org > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > > > > > > > > -- > > Carsten Bock > > Schomburgstr. 80 > > 22767 Hamburg > > Germany > > > > Mobile +49 179 2021244 > > Home +49 40 34927217 > > Büro (Verl) +49 5246 801427 > > Fax +49 40 34927218 > > mailto:carsten@bock.info > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >

13 years, 10 months

fix_nated_sdp()

by Bernhard Suttner

Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) && has_sdp() && nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner

13 years, 10 months

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sr-users January 2011