[SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
i got mozilla to generate sdp with sendrecv, but still rtpengine does
not replace 0.0.0.0 address on o and c lines. why?
-- juha
Dec 19 10:20:18 box /usr/bin/sip-proxy[5841]: INFO: =
rtpengine_offer(ICE=force replace-session-connection replace-origin
via-branch=1)
Dec 19 10:20:18 box rtpengine[20171]: Got valid command from 127.0.0.1:39586:
offer - { sdp: v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4
0.0.0.0#015#012s=SIP Call#015#012t=0
0#015#012a=ice-ufrag:f5511cb8#015#012a=ice-pwd:1d062e6b56db4f738c7613bc97086194#015#012a=fingerprint:sha-256
45:CA:2E:A5:12:4E:80:0C:AD:9D:11:94:C5:2D:ED:15:55:FD:83:70:91:59:3D:57:3D:F9:A5:77:74:85:BD:E8#015#012m=audio
9 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109
opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8 ...
Dec 19 10:20:18 box rtpengine[20171]: ... 000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012a=extmap:1
urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=rtcp-mux#015#012a=candidate:0
1 UDP 2122252543 192.168.1.52 42190 typ host#015#012a=candidate:0 2 UDP
2122252542 192.168.1.52 54801 typ host#015#012a=candidate:1 1 UDP 1686110207
192.98.100.128 42190 typ srflx raddr 192.168.1.52 rport
42190#015#012a=candidate:1 2 UDP 1686110206 192.98.100.128 54801 typ srflx
raddr 192.168.1.52 rport 54801#015#012, ICE: force, replace: [
session-connection, origin ], call-i ...
Dec 19 10:20:18 box rtpengine[20171]: ... d: lr92g1vkbkvmgdt9m1dg,
via-branch: z9hG4bK2313203, received-from: [ IP4, 192.98.100.128 ],
from-tag: d03ug77pi6, command: offer }
Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Creating new call
Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports
8056..8057 for media relay
Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports
8012..8013 for media relay
Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Returning to SIP
proxy: d3:sdp725:v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4
0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-lite#015#012m=audio 8056
RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109
opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012a=rtcp:8057#015#012a=rtcp-mux#015#012a=crypto:1
AES_CM_128_HMAC_SHA1_80
inline:8FwYLdKkGvVH4Ykc+kTgrFN1Cwz0KZVpT2JK7+g0#015#012a=setup:actpass#015#012a=fingerprint:sha-1
ED:DE:59:BA:F7 ...
Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] ...
:F8:78:9D:1D:78:42:AA:00:6C:88:85:A4:2B:36:42#015#012a=ice-ufrag:4dmKpmV3#015#012a=ice-pwd:75jY5mekvVULGZnnB9mJmHQoVa9i#015#012a=candidate:bVSyZZLCFRG94R1D
1 UDP 2130706431 192.26.111.29 8056 typ
host#015#012a=candidate:bVSyZZLCFRG94R1D 2 UDP 2130706430 192.26.111.29 8057
typ host#015#0126:result2:oke
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[SR-Users] Unregister is failing
Hi All, I am running ims servers(pcscf,scscf,icscf and hss) as part of kamailio proxy. And I am trying to register and unregister the end-points, with help osip lib. But I see the below ERROR message. ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio/pcscf/kamailio.cfg] l=908 a=25 n=t_check_status Please help me to fix the issue. Thanks and Regards, Anil ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Can't start Kamailio with both db_postgres and tls
On 18/12/14 22:09, James Cloos wrote: DM == Daniel-Constantin Mierla mico...@gmail.com writes: DM The question would be more specific to the error message printed from DM postgres client library: DM FATAL: no pg_hba.conf entry for host 129.240.1.1, user DM foo_test_user, database foo_test, SSL off DM Is it something that is documented somewhere or maybe some web search DM can indicate the reasons why it happens? His initial post mentioned that they require ssl for the pg tcp sockets; the error about pg_hba just confirms that. Can you elaborate? Otherwise I don't see what's the role of this reply, because that was clear they want want tls for postgres. The error message says 'no pg_hba.conf entry for host ...' -- sounds like something missing in postgres client/server configuration. Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIP Fragments
That's how I ended up going. It's working now. Thanks. On Thu, Dec 18, 2014 at 4:11 PM, James Cloos cl...@jhcloos.com wrote: MS == Marc Soda ms...@coredial.com writes: MS I'm having a problem reassembling UDP packets on my Asterisk servers after MS passing through Kamailio You could try having the kama-ast socket use tcp. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- Marc Soda Sr. Systems Architect, Production Systems // *CoreDial, LLC* // coredial.com // [image: Twitter] https://twitter.com/coredial [image: LinkedIn] http://www.linkedin.com/company/99631 [image: Google Plus] https://plus.google.com/104062177220750809525/posts [image: Blog] http://success.coredial.com/blog 751 Arbor Way, Hillcrest I, Suite 150, Blue Bell, PA 19422 *P: *215.297.4400 x203 // *F: *215.297.4401 // *E: * ms...@coredial.com [image: Success Starts Here. Check out our blog and subscribe today!] http://cta-service-cms2.hubspot.com/cs/c/?cta_guid=59f8a888-5f9f-4320-bd00-59436c5af213placement_guid=9ea684ae-6f3c-407c-a8f7-1c93aa481db6portal_id=210539redirect_url=a/M/ef3qXLrO9PM9nt4KAyUqKgEk1bjS%2BVrjJ9WLfaFcK2fhsQYM3JV0AXyMHtXQEFZiDnBKD38%3Div=4s02rvvxZAY%3Dhsutk=canon=http%3A%2F%2Fwww.stratusinteractive.com%2Fcoredial%2Femail%2Fcoredial-signature.html The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
On 12/19/14 03:32, Juha Heinanen wrote: i got mozilla to generate sdp with sendrecv, but still rtpengine does not replace 0.0.0.0 address on o and c lines. why? Because 0.0.0.0 means steam is on hold and so should be left in place. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Accounting: syslog vs db_flatfile
Hi, DanB! Kamailio has radius accounting too. -- WBR, Victor I use FREE operation system: 3.15.9- GNU/Linux up 2 weeks, 1 day, 5 hours, 22 minutes ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
Richard Fuchs writes: On 12/19/14 03:32, Juha Heinanen wrote: i got mozilla to generate sdp with sendrecv, but still rtpengine does not replace 0.0.0.0 address on o and c lines. why? Because 0.0.0.0 means steam is on hold and so should be left in place. what i understand from rfc3264 is that 0.0.0.0 address in initial invite does not mean that the call is on hold. 0.0.0.0 is used because webrtc client does not know its ip address: RFC 2543 [10] specified that placing a user on hold was accomplished by setting the connection address to 0.0.0.0. Its usage for putting a call on hold is no longer recommended, since it doesn't allow for RTCP to be used with held streams, doesn't work with IPv6, and breaks with connection oriented media. However, it can be useful in an initial offer when the offerer knows it wants to use a particular set of media streams and formats, but doesn't know the addresses and ports at the time of the offer. also, if the call would on hold, there would be sendonly attribute in the sdp, which is not the case in my example, which had sendrecv. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
On 12/19/14 09:33, Juha Heinanen wrote: Richard Fuchs writes: On 12/19/14 03:32, Juha Heinanen wrote: i got mozilla to generate sdp with sendrecv, but still rtpengine does not replace 0.0.0.0 address on o and c lines. why? Because 0.0.0.0 means steam is on hold and so should be left in place. what i understand from rfc3264 is that 0.0.0.0 address in initial invite does not mean that the call is on hold. 0.0.0.0 is used because webrtc client does not know its ip address: RFC 2543 [10] specified that placing a user on hold was accomplished by setting the connection address to 0.0.0.0. Its usage for putting a call on hold is no longer recommended, since it doesn't allow for RTCP to be used with held streams, doesn't work with IPv6, and breaks with connection oriented media. However, it can be useful in an initial offer when the offerer knows it wants to use a particular set of media streams and formats, but doesn't know the addresses and ports at the time of the offer. also, if the call would on hold, there would be sendonly attribute in the sdp, which is not the case in my example, which had sendrecv. Yes I understand, but 1) the mechanism of using 0.0.0.0 to put a call on hold must remain operational and intact for those clients which use it, and 2) if the offering client sends 0.0.0.0 in the SDP, then the rewritten SDP should also contain 0.0.0.0, no matter what the purpose behind it is. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
Richard Fuchs writes: Yes I understand, but 1) the mechanism of using 0.0.0.0 to put a call on hold must remain operational and intact for those clients which use it, and 2) if the offering client sends 0.0.0.0 in the SDP, then the rewritten SDP should also contain 0.0.0.0, no matter what the purpose behind it is. So with current implementation of rtpengine there is no hope if Firefox webrtc client needs rtpengine services? If so, would it be possible have a new flag that tells rtpengine to replace also 0.0.0.0 address if there is no sendonly attribute in sdp? -- Juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
On 12/19/14 10:02, Juha Heinanen wrote: Richard Fuchs writes: Yes I understand, but 1) the mechanism of using 0.0.0.0 to put a call on hold must remain operational and intact for those clients which use it, and 2) if the offering client sends 0.0.0.0 in the SDP, then the rewritten SDP should also contain 0.0.0.0, no matter what the purpose behind it is. So with current implementation of rtpengine there is no hope if Firefox webrtc client needs rtpengine services? If so, would it be possible have a new flag that tells rtpengine to replace also 0.0.0.0 address if there is no sendonly attribute in sdp? I don't see how it would make a difference. If Firefox sends 0.0.0.0 and rtpengine replaces it with its own address, then the receiving client can send media to rtpengine, but rtpengine would have nowhere to forward it to. After the answer, ICE processing may commence and determine an IP address, after which I expect Firefox to send an updated offer with the address filled in. At this point, media should start to flow no matter what. I'm not sure how much of a valid use-case this is, or if it'd be just a Firefox-specific workaround, but my all means, give it a try and see if it makes a difference. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Media trouble with kamailio/rtpengine
I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser - kamailio/rtpengine - asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 Thanks. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
Richard Fuchs writes: I don't see how it would make a difference. If Firefox sends 0.0.0.0 and rtpengine replaces it with its own address, then the receiving client can send media to rtpengine, but rtpengine would have nowhere to forward it to. After the answer, ICE processing may commence and determine an IP address, after which I expect Firefox to send an updated offer with the address filled in. At this point, media should start to flow no matter what. I'm not sure how much of a valid use-case this is, or if it'd be just a Firefox-specific workaround, but my all means, give it a try and see if it makes a difference. i found this on jssip mailing list: Just a side note; FF34 does offer the value '0.0.0.0' as the media connectiom address, and the value 9 as the media port. This way it announces the support for ICE Trickle. Media servers not aware of this will take it as a 'hold' request and wont send media to the peer. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Media trouble with kamailio/rtpengine
Even stranger, I get a media stream back to the browser when I use Chrome (the first was with Firefox), but I still hear nothing. Also I get errors like this in the log: SRTP output wanted, but no crypto suite was negotiated Full output: https://gist.github.com/marcantonio/6c5414aa931a8f1c0072 On Fri, Dec 19, 2014 at 10:47 AM, Marc Soda ms...@coredial.com wrote: I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser - kamailio/rtpengine - asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 Thanks. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Initial REGISTER request/response in 3rd Party Register
Hello, I'm using Kamailio with a SIP Application Server, when a user registers on the IMS Core a 3rd Party REGISTER request is sent to the application server to start some logic. I'm trying to add in the 3rd party register request body the initial REGISTER request sent by user device, and the 200 OK response sent back by Kamailio. Do you think it is possible to configure Kamailio to get this behaviour ? Thank you! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpengine returns 0.0.0.0 even with sendrecv in offer
On 12/19/14 11:39, Juha Heinanen wrote: Richard Fuchs writes: I don't see how it would make a difference. If Firefox sends 0.0.0.0 and rtpengine replaces it with its own address, then the receiving client can send media to rtpengine, but rtpengine would have nowhere to forward it to. After the answer, ICE processing may commence and determine an IP address, after which I expect Firefox to send an updated offer with the address filled in. At this point, media should start to flow no matter what. I'm not sure how much of a valid use-case this is, or if it'd be just a Firefox-specific workaround, but my all means, give it a try and see if it makes a difference. i found this on jssip mailing list: Just a side note; FF34 does offer the value '0.0.0.0' as the media connectiom address, and the value 9 as the media port. This way it announces the support for ICE Trickle. Media servers not aware of this will take it as a 'hold' request and wont send media to the peer. Oh dear, another IETF draft... Well that certainly explains, thanks. Doesn't look like Firefox is quite finished with it yet though, as ice-options=trickle isn't given. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Media trouble with kamailio/rtpengine
On 12/19/14 10:47, Marc Soda wrote: I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser - kamailio/rtpengine - asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 You've caught the same thing as Juha did just earlier, Firefox is doing something new called Trickle ICE, which at the moment breaks communications with endpoints not supporting it (such as rtpengine). The second call you posted seems fine. The error you're seeing is because RTP was received before DTLS was established and so is expected. You can try --dtls-passive as a possible fix. Media should start to flow after DTLS gets established though, and according to the logs, media was indeed seen in both directions. Try tcpdump to confirm. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Media trouble with kamailio/rtpengine
Thanks for the response. You're right, the media stream is making it all the way back to my PC, I just don't hear anything. And yes, my speakers are turned up... I'm not sure what to try next... On Fri, Dec 19, 2014 at 12:31 PM, Richard Fuchs rfu...@sipwise.com wrote: On 12/19/14 10:47, Marc Soda wrote: I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser - kamailio/rtpengine - asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 You've caught the same thing as Juha did just earlier, Firefox is doing something new called Trickle ICE, which at the moment breaks communications with endpoints not supporting it (such as rtpengine). The second call you posted seems fine. The error you're seeing is because RTP was received before DTLS was established and so is expected. You can try --dtls-passive as a possible fix. Media should start to flow after DTLS gets established though, and according to the logs, media was indeed seen in both directions. Try tcpdump to confirm. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Can't start Kamailio with both db_postgres and tls
DM == Daniel-Constantin Mierla mico...@gmail.com writes: DM Can you elaborate? Otherwise I don't see what's the role of this reply, DM because that was clear they want want tls for postgres. Apologies that I wasn't clear. DM The error message says 'no pg_hba.conf entry for host ...' -- sounds DM like something missing in postgres client/server configuration. It is. The important part is the no_ssl part; that just confirms what the poster wrote in his first message, that they limit connections to their pg to require ssl when connecting via tcp from host 129.240.1.1 as foo_test_user. There is one issue I didn't spot at first, though. In the db name section of the error: FATAL: no pg_hba.conf entry for host 129.240.1.1, user foo_test_user, database foo_test, SSL off there is an extra space, foo_test vs foo_test. That might turn out to be the entire issue. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] The Subscriber table VS the Location table ?!
Hi Dears, I'm a little bit confused about the difference between the Subscriber table and the Location table.I read that the Location table is used for persistent user registration BUT i did NOT configured the Location table and can get persistent user registration with the Subscriber table ! So can anyone explain the difference please ? Best Regards and Thanks in Advance. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] The Subscriber table VS the Location table ?!
The users in subscriber table are the actual users who are allowed to register to your SIP service. This is where kamailio gets the authentication information, e.g. username and password etc. The location table is where kamailio stores currently registered i.e. online users. Obviously the records in location table are subset of subscriber table. You do not need to maintain the location table in normal cases as kamailio takes care of that itself. Thank you. On Fri, Dec 19, 2014 at 11:35 PM, Mahmoud Ramadan Ali cisco.and.more.b...@gmail.com wrote: Hi Dears, I'm a little bit confused about the difference between the Subscriber table and the Location table.I read that the Location table is used for persistent user registration BUT i did NOT configured the Location table and can get persistent user registration with the Subscriber table ! So can anyone explain the difference please ? Best Regards and Thanks in Advance. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] The Subscriber table VS the Location table ?!
On 19 Dec 2014, at 23:35, Mahmoud Ramadan Ali cisco.and.more.b...@gmail.com wrote: Hi Dears, I'm a little bit confused about the difference between the Subscriber table and the Location table.I read that the Location table is used for persistent user registration BUT i did NOT configured the Location table and can get persistent user registration with the Subscriber table ! So can anyone explain the difference please ? The subscriber table is for AUTHENTICATION. It's not for registrations specifically, as you can authenticate anything but CANCEL. The location table is by default kept in memory. When you reboot Kamailio it's gone. If you add a database we can store registrations there in many different ways - as a backup of in-memory data, instead of in-memory or in sync with in-memory. It's two very different tables. /O ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
