sr-users December 2010

sr-users@lists.kamailio.org

63 participants
114 discussions

RTP to SRTP bridge
by Dominguez Jover, Ricardo 22 Dec '10

22 Dec '10

Hi everybody, I'm trying to deploy an scenario where all calls in the LAN are encrypted with SRTP, so I've forced my sip softphones to "mandatory SRTP". As in the Internet not all providers and clients have SRTP enabled, I couldn´t communicate them in this way, so I would need some kind of border element to bridge RTP and SRTP calls. Is there any bridge/proxy who can receive rtp packets and convert them into srtp and viceversa? I've been reading RTPproxy and Media proxy docs and also some forums, but I didn´t find anything useful. Regards, Ricardo Dominguez

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Re: [SR-Users] 404 Not Found - If User Not Registered
by Klaus Darilion 22 Dec '10

22 Dec '10

Am 22.12.2010 13:29, schrieb David J.: > Klaus; > > The docs don't seem so clear as to which state is User Exists and Not > Registered vs No User Exists; You can not tell if a user exists by querying the location table - you can only tell if a user is registered. To find out if a user exists you would have to query the subscriber table. regards klaus

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RLS issue
by "Andrés S. García Ruiz" 22 Dec '10

22 Dec '10

Hi everybody, I'm using the Kamailio RLS+Presence module integrated with OpenIMSCore. All the servers are running properly but when sending a RLS subscribe. The flow of SIP messages is the following: A.1) Watcher ---- SUBSCRIBE (rls-services uri) ---> RLS SUBSCRIBE sip:restricted_areas@open-ims.test SIP/2.0 Call-ID: dcKTUqpGi8lsPuQzqI0m922XEmSQpAxU36XScBNuu1A. CSeq: 1 SUBSCRIBE From: "restricted_areas" <sip:restricted_areas@open-ims.test>;tag=11afb48e To: <sip:restricted_areas@open-ims.test> Via: SIP/2.0/TCP 155.54.190.166:8060;rport;branch=z9hG4bK-d8754z-29095b2cee722e34-1---d8754z- Max-Forwards: 70 Event: presence Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/auth-policy+xml Expires: 30000 Contact: <sip:restricted_areas@155.54.190.166:8060> Supported: eventlist Route: <sip:orig@scscf.open-ims.test:6060;lr> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Thu_Dec_16_10-48-31_CET_2010 Content-Length: 0 A.2) RLS ---- 200 Ok ----> Watcher B.1) RLS ---- SUBSCRIBE (presentity in the rls-services user's list) ---> I-CSCF SUBSCRIBE sip:restricted_areas@open-ims.test SIP/2.0 Record-Route: <sip:mo@scscf.open-ims.test:6060;lr> Route: <sip:ciervo.inf.um.es:5060;lr>, <sip:iscmark@scscf.open-ims.test:6060;lr;s=1;h=0;d=0;a=7369703a726573747269637465645f6172656173406f70656e2d696d732e74657374> Record-Route: <sip:mo@pcscf.open-ims.test:4060;lr> Call-ID: dcKTUqpGi8lsPuQzqI0m922XEmSQpAxU36XScBNuu1A. CSeq: 1 SUBSCRIBE From: "restricted_areas" <sip:restricted_areas@open-ims.test>;tag=11afb48e To: <sip:restricted_areas@open-ims.test> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bKfc46.88774fc7.0 Via: SIP/2.0/UDP 155.54.210.134:4060;branch=z9hG4bKfc46.8dfcbef7.0;i=2 Via: SIP/2.0/TCP 155.54.190.166:8060;rport=42816;branch=z9hG4bK-d8754z-29095b2cee722e34-1---d8754z- Max-Forwards: 15 Event: presence Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/auth-policy+xml Expires: 30000 Contact: <sip:restricted_areas@155.54.190.166:8060> Supported: eventlist Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite IMS-OSGi-Client 0.1 CVS-Thu_Dec_16_10-48-31_CET_2010 Content-Length: 0 P-Asserted-Identity: <sip:restricted_areas@open-ims.test> P-Charging-Vector: icid-value="P-CSCFabcd000000004d09e21300000005";icid-generated-at=155.54.210.134;orig-ioi="open-ims.test" B.2) I-CSCF ---- QUERY ----> HSS B.3) I-CSCF ---- SUBSCRIBE ----> S-CSCF SUBSCRIBE sip:testuser01@open-ims.test SIP/2.0 Route: <sip:scscf.open-ims.test:6060> Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 16 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist B.4) S-CSCF ---- SUBSCRIBE ----> P-CSCF SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 SIP/2.0 Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> Route: <sip:term@pcscf.open-ims.test:4060;lr> Via: SIP/2.0/UDP 155.54.210.135:6060;branch=z9hG4bK07df.8a35e4f3.0 Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 15 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist P-Called-Party-ID: <sip:testuser01@open-ims.test> B.5) P-CSCF ---- SUBSCRIBE ----> Presentity??? SUBSCRIBE sip:testuser01@155.54.190.245:8060;rinstance=9b7761b4bcaa4bd0 SIP/2.0 Record-Route: <sip:mt@pcscf.open-ims.test:4060;lr> Record-Route: <sip:mt@scscf.open-ims.test:6060;lr> Via: SIP/2.0/TCP 155.54.210.134:4060;branch=z9hG4bK07df.baa7cf24.0 Via: SIP/2.0/UDP 155.54.210.135:6060;received=155.54.210.135;rport=6060;branch=z9hG4bK07df.8a35e4f3.0 Via: SIP/2.0/UDP 155.54.210.136;branch=z9hG4bK07df.3688b985.0 Via: SIP/2.0/UDP 155.54.190.245;branch=z9hG4bK07df.964e0ba7.0 To: sip:testuser01@open-ims.test From: sip:restricted_areas@open-ims.test;tag=533cb9e91f4b999cf76861cbb9ed54ed-32d5 CSeq: 10 SUBSCRIBE Call-ID: 7fd8dfdd-21694(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.2.0-dev1 (i386/linux)) Max-Forwards: 14 Event: presence Contact: <sip:155.54.190.245:5060;transport=udp> Expires: 10810 Max-Forwards: 70 Support: eventlist P-Called-Party-ID: <sip:testuser01@open-ims.test> The step B.5 is sent directly to the presentity testuser01. Instead of that, I thought that message was suppose to be sent to the presence server in the URI that is shown in Contact header. Regards, Andrés.

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kamailio restart and TLS ( relay_to_tls() )
by Dominguez Jover, Ricardo 22 Dec '10

22 Dec '10

Hi everybody, Since I implemented Kamailio 3.1 with TLS I've found a strange behavior. That is, with some clients (Bria and Blink) registered, if I restart Kamailio, then when the clients re-register the strange behaivour happens. This behavior consist on receiving calls, it took about 15 seconds to receive the first tone since the call was made. I made the following modification in the "route[Relay]" config. The reason is I wanted my gateway and Kamailio to make signaling by TLS. Without this modification the signaling was unencrypted (SIP): route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } # Se comunica con el GWa traves de TLS if(!( ($od=~"mydomain.com") && ( ($rU=~"[a-z]{3,20}$") || ($rU=~"^xx[0-9][0-9]$") ) ) ) { ### If I'm calling a PBX extension do the signaling by TLS with the gateway (Cisco 2811) if (!t_relay_to_tls()) { sl_reply_error(); } } else if { if (!t_relay()) { sl_reply_error(); } } exit; } The rest of functionalities are working really fine. Any idea about what is happening? Cheers! Ricardo Domínguez

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404 Not Found - If User Not Registered
by David J. 22 Dec '10

22 Dec '10

if I do lookup() What case does lookup return if entry exists but user not registered? lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; should I just wrap this like this: if(registered("location")){ lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; } }; The purpose I ask this rather simple question is because I want to redirect to voicemail if user is not registered. In the default script we have this case; failure_route[FAIL_ONE] { # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } But it seems this case is never met; Because this code lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; Seems to send a 404 response to the UA rather than go to failure route? Maybe I am wrong... Any advice would be appreciated.

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Re: [SR-Users] SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 22 Dec '10

22 Dec '10

Trying the other registrations takes 10 seconds extra for each registration above the first: registrations time before phone starts ringing --------------- -------------------------------- 1 3s 2 13s 3 23s 4 33s 5 43s --------------- -------------------------------- Does anyone see a way to limit the number of registrations to allow only 1 per User-Agent/IP-address combination? Op 22 dec 2010, om 02:08 heeft David J. het volgende geschreven: > Look at the registration module; > > There is a max_contacts param; > > Its fine if the UA re-registers; Its when SER sends out to the UA; Its only going to reach the one which is available; > > It may try the others but they obviously wont be reachable. >

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Re: [SR-Users] SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 22 Dec '10

22 Dec '10

Regarding the business rule, can you explain more about > YOu can specify max contacts on the registration of an endpoint For instance, how and where is the max contacts specified? There still might be an issue, as it is pointless to store multiple registrations for a single user-agent that re-regisers (without un-registering) using the same IP address, like below: bluesip/user,sip:pb@86.8.210.156:49569;rinstance=991CCEE9;transport=tcp2010-12-22 01:26:360.50 bluesip/user,sip:pb@86.8.210.156:49584;rinstance=991CCEE9;transport=tcp2010-12-22 01:33:250.50 bluesip/user,sip:pb@86.8.210.156:49585;rinstance=991CCEE9;transport=tcp2010-12-22 01:35:330.50 bluesip/user,sip:pb@86.8.210.156:49586;rinstance=991CCEE9;transport=tcp2010-12-22 01:37:430.50 bluesip/user,sip:pb@86.8.210.156:49587;rinstance=991CCEE9;transport=tcp2010-12-22 01:39:510.50 bluesip/user,sip:pb@86.8.210.156:49588;rinstance=991CCEE9;transport=tcp2010-12-22 01:42:000.50 Only the last line is the active one. The first five registrations are old and overridden by the newer registration. For instance: ports 49569, 49584, 49585, 49586 and 49587 are no longer in the NAT routing table, only 49588 is. Op 22 dec 2010, om 01:03 heeft David J. het volgende geschreven: > Its not an issue; > > AG Projects may allow multiple registrations from the SAME IP whereas the other company may not; > > Thats a business rule; Not a bug; YOu can specify max contacts on the registration of an endpoint. > > Its not a bug. > > > > On 12/21/10 6:58 PM, cerieljacobs wrote: >> Bluesip.net uses "Sip EXpress router (0.9.7 (i386/linux))" >> and on client registration it returns >> "SIP/2.0 403 Max Registrations per IP exceeded" >> due to some bug/error/failure in the version 1.4 Acrobits Business softphone user-agent >> to unregister correctly. >> >> Strange is that this never happens with accounts on AG Projects, "OpenSIPS (1.4.5-notls (x86_64/linux))" server. This server software seems to remove old registrations for the same user-agent from the same ip address. >> >> What should be changed in Sip EXpress router, >> to store only one single registration per user-agent/IP combination? >> >> Thanks,Ceriel >> _______________________________________________ >> 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 >> >

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SER0.9.7: How to prevent: 403 Max Registrations per IP exceeded
by cerieljacobs 21 Dec '10

21 Dec '10

Bluesip.net uses "Sip EXpress router (0.9.7 (i386/linux))" and on client registration it returns "SIP/2.0 403 Max Registrations per IP exceeded" due to some bug/error/failure in the version 1.4 Acrobits Business softphone user-agent to unregister correctly. Strange is that this never happens with accounts on AG Projects, "OpenSIPS (1.4.5-notls (x86_64/linux))" server. This server software seems to remove old registrations for the same user-agent from the same ip address. What should be changed in Sip EXpress router, to store only one single registration per user-agent/IP combination? Thanks,Ceriel

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SBC on kamailio
by Henry Dogger 21 Dec '10

21 Dec '10

Hi all, We have a kamailio installation for our voip system and would like to add a SBC. We found the freeswitch possibility: http://wiki.freeswitch.org/wiki/SBC_Setup But in this case, 302 Redirect SIP is used, we would like the SBC to act as a pass-through controller, just to be a gateway for SIP clients from outside. Is something like this possible? And will registrations be handled correctly since all SIP messages will be on port 5060. Thanks in advance. Kind regards, Henry Dogger Telecats BV

3 7

Recursive failure_route after CFNA
by Igor Potjevlesch 21 Dec '10

21 Dec '10

Hello all, I have setup a mechanism to forward outgoing calls to another gateway if the first one replies with 5xx. I have simply used ‘t_check_status’ in the failure_route. This works. But, with a CFNA it doesn’t work : the incoming call is treated by SER (0.9.9). After the timer has expired the call goes to ‘failure_route’ for the first time to setup the CFNA with, finally, an ‘append_branch’. At this moment, the first gateway replies with 5xx and the call never goes to ‘failure_route’. I can match the 5xx within the ‘onreply_route’, but that’s all. So, instead of trying another gateway, the initial incoming calls terminate on the voicemail. Can anyone help me? Regards, Igor.

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sr-users December 2010