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directed media problem when video is included in sdp
by Juha Heinanen 20 Sep '11

20 Sep '11

when sems receives voice call invite from ua behind nat that has direction:active in its sdp, like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1. 68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Attribute (a): direction:active sems includes direction:passive in 200 OK and starts to correctly send rtp packets to the ip address/port of received rtp packets. however, then sems receives a voice+video call invite like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1.68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Description, name and address (m): video 8600 RTP/AVP 34 Bandwidth Information (b): AS:300 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:34 H263/90000 Media Attribute (a): direction:active it does not include direction:passive in 200 OK and keeps on sending rtp packets to 192.168.1.68 even after receiving first voice rtp packet from public ip of the ua. is this a bug in sems or is it a bug in sr nathelper_k fix_nated_sdp("1") implementation, i.e., does that function place direction:active line to only one place, when it should place it in two places (in both voice and video media descriptions)? -- juha

3 4

Module db_text
by Bruno Bresciani 20 Sep '11

20 Sep '11

Hi All, I'm starting kamailio version 3.1.2 with db_text module but when it tries to load route table, the following error is generated: * "ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe" * Someone Can Help me to understand why this error is happening? Below is part of kamaio's trace. Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:208]: _tbc->name: route Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:209]: _tbc->nrcols: 9 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:210]: _nc: 8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:238]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_res.c:69]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0x82e23f0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x82e3798 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:165]: allocate 32 bytes for result types at 0x82e37c0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[0] at 0x82e2438 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[1] at 0x82e2390 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[2] at 0x82e23d0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[3] at 0x82e23e0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[4] at 0x82e37e8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[5] at 0x82e37f8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[6] at 0x82e3808 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[7] at 0x82e3818 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:184]: allocate 8 bytes for rows at 0x82e3828 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_row.c:119]: allocate 160 bytes for row values at 0x82e3838 Sep 19 16:13:01 sswpst00 kamailio: ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe Best Regards

2 1

Re: [SR-Users] rtpproxy VS Turn
by Mikael Williams 19 Sep '11

19 Sep '11

> > ICE is not supported as I select the Media Proxy (which is TURN) manually > in some Nat cases...Even note that I do not have peer to peer calls between > 2 endpoints, just have VOIP to PSTN calls...All my VOIP customers supports > STUN/TURN but without ICE > > One more thing: I would like to ask how rtp proxy help to bypass NAT else > than working as relay server...Does it have any other job like find the NAT > type or check blocked port or something? > > Regards > > > On Fri, Sep 16, 2011 at 5:50 PM, Klaus Darilion < > klaus.mailinglists(a)pernau.at> wrote: > >> STUN alone is not 100% sufficient, only ICE+TURN is reliable. Thus, only >> if all your clients support ICE+TURN then you can drop rtpproxy. >> >> klaus >> >> >> On 16.09.2011 16:47, Mikael Williams wrote: >> >>> Dear All, >>> >>> I would like to ask please if and in case I'm using STUN/TURN for Nat >>> traversal to drp the rtpproxy from my solution as NAT traversal and >>> relay server as I'm using SYUN for NAT traversal and TURN as relay in >>> case of Symmetric NAT >>> >>> Regards >>> >>> >>> ______________________________**_________________ >>> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >

2 1

Re: [SR-Users] sipp to kamailio doesn't match ACK [SOLVED]
by Uri Shacked 19 Sep '11

19 Sep '11

Hi, I got the same problem. Do you have the XML SIPp file for example? 10x Uri

1 0

Dialog module showing calls that have already been terminated
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Hello I have been facing an issue where the dialog module is showing calls as being active when in fact the call has already been completed long ago and this is giving wrong number of concurrent calls on our SNMP work station (CACTI) when polling the data from Kamailio. I realized this only started occurring after i upgraded from 3.1.2 to 3.1.5, has anyone experienced the same issue? my config: ======== loadmodule "dialog.so" . . . . # ----------------- DIALOG MODULE PARAMETERS----------------------------------# modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_extra_hdrs", "NULL") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 60) modparam("dialog", "db_fetch_rows", 500) modparam("dialog", "table_name", "dialog") modparam("dialog", "from_uri_column", "from_uri") modparam("dialog", "from_tag_column", "from_tag") modparam("dialog", "to_uri_column", "to_uri") modparam("dialog", "to_tag_column", "to_tag") modparam("dialog", "h_id_column", "hash_id") modparam("dialog", "h_entry_column", "hash_entry") modparam("dialog", "state_column", "state") modparam("dialog", "start_time_column", "start_time") modparam("dialog", "timeout_column", "timeout") modparam("dialog", "sflags_column", "sflags") modparam("dialog", "bridge_controller", "sip:controller@kamailio.org") modparam("dialog", "default_timeout", 7200) . . . . route { # MANAGE ALL DIALOGS #=================================================== if (is_method("INVITE")) { if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = 12; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); } Thanks Phillip

2 3

kamailio debian/ubuntu support
by Jon Bonilla 16 Sep '11

16 Sep '11

Hi all with the release of kamailio 3.1.5, I've done a small change to the debian repositories: Nightly builds of kamailio_3.0 branch have been dropped and the repository has been erased. There is no development in this branch and it has no sense to keep nightly builds for it. Those (few) people using kamailio_3.0-nightly repositories are advised to move their sources.list to the stable 3.0 repository of upgrade to 3.1 or master branches. kamailio 3.0 will continue to be packaged if any new stable version is released for etch, lenny, squeeze and lucid. For 3.1 branch, nightly builds and stable repos will be provided for lenny, lucid and squeeze When the time for 3.2 (master branch at the moment) comes, I'll add Wheezy support. Not sure about Ubuntu distro support. Should we add support to Ubuntu 11.10 or wait until the lts 12.04 is released? Any opinions welcomed cheers, Jon

4 4

rtpproxy VS Turn
by Mikael Williams 16 Sep '11

16 Sep '11

Dear All, I would like to ask please if and in case I'm using STUN/TURN for Nat traversal to drp the rtpproxy from my solution as NAT traversal and relay server as I'm using SYUN for NAT traversal and TURN as relay in case of Symmetric NAT Regards

2 1

Communication from IPv6 to Ipv4
by Loïc Fofana 16 Sep '11

16 Sep '11

My architecture is described below: TDM device----------- Call server[IPv4]---------------[IPv4]Kamailio[IPv6]------------[IPv6]Asterisk---------[IPv6]Linphone * * * * * * * * * * * * Media Server Rtpproxy Media Server and Rtpptoxy carry the media. Call Server carry only signaling. I have succeeded to establish audio communication between an IPv6 domain & an IPv4 domain i.e. : from an TDM device to a Linphone. However, When i try to establish a call from (IPv6 to IPv4), i.e. from Linphone to TDM device, the signaling is OK, but there's no voice. Because, Rtpproxy forward the media to the Call server instead the Media Server. Hence, there can't be voice. I have been stucking on this for a couple of months. I will be very glad if somebody have any idea. best regards!

2 1

Re: [SR-Users] Kamailio terminating after upgrading from 3.1.2 to 3.1.4
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Dear Marius Thank you for your prompt reply. I apologise, As i am a newbie could you please explain how to produce the below output that you require? Thanking you in advance for your help! Regards Phillip > Hello > Hello. You should have a coredump . Can you please send the trace (bt full) as seen with gdb ? Thanks Marius > Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. > The kamailio process keeps on terminating every now and then, this is > what i was able to retrieve from my logs. > > > Sep 14 12:52:31 SipProxy1 kernel: [71448.062089] __ratelimit: 15 > callbacks suppressed > Sep 14 12:52:31 SipProxy1 kernel: [71448.062096] kamailio[13963]: > segfault at 14 ip 0816452a sp bfedc940 error 4 in kamailio[8048000+1d9000] > > > Does anyone have an idea why this is happening? > > Your help is much appreciated! > > Regards > Phillip

5 9

(no subject)
by Loïc Fofana 16 Sep '11

16 Sep '11

This is my scenario: Call server -------------(IPV4)Kamailio(IPV6)----------Asterisk(IPV6)------Linphone(IPV6) * * * Media server

2 1

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