sr-users August 2019

sr-users@lists.kamailio.org

65 participants
97 discussions

set_rtpengine_set question
by Juha Heinanen 24 Aug '19

24 Aug '19

When set_rtpengine_set() is called with two parameters before rtpengine_offer(), e.g. set_rtpengine_set("1", "2"); rtpengine_offer(); it was not clear to me from rtpengine README, if the sets need to reversed when set_rtpengine_set() is called before rtpengine_answer(), i.e., set_rtpengine_set("2", "1"); rtpengine_answer(); or can the order of the sets be the same as it was in case of rtpengine_offer()? -- Juha

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Dealing with optional SRTP
by Mark Boyce 24 Aug '19

24 Aug '19

Hi all (Using Kamailio + RTPEngine, both current stable versions) Just wondering what current best practice is for dealing with “optional SRTP” such as those generated by Yealink devices. (I know this breaks the RFC but lots of things seem to use this method.) Where the device SDP message looks like; m=audio 50004 RTP/AVP 8 97 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhJOWOXPuzMeBmrbx2wAKohuveAvZeMEJTzfE7GT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:yF+EhR3RtxSDCGtLCnEAl3Zrdut4b8cfFQP+hFqf An unencrypted RTP/AVP media line but with crypto lines showing that the device will support SRTP encryption but it is not mandatory. The system of offer SRTP and try and catch on not supported/acceptable here fail feels a little clumsy and relies on how gracefully the device sends “not acceptable here”. I’m wondering if I just short cut it and say that if a device offers a acceptable crypto line then Kamailio sends back a SAVP response. Naturally this question is then closely followed by how to do optional SRTP invites originating from Kamailio without just directly manipulating lines? Feels like there should be an RTPEngine setting for Optional? RTP/AVP+SAVP ? Thoughts? Mark

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Kamailio as an Outbound Proxy - Strange Sip Flow
by Diogo Hartmann 23 Aug '19

23 Aug '19

Hello Kamailio experts, We have a Kamailio as outbound proxy in front of Asterisks PjSIP. There is a strange behavior after 200 OK: - ACK comes from Asterisk Public IP instead of coming from the internal IP address. - BYE can not be processed (404 not here) SNGREP print attached. Has anyone ever experienced this? I can provide more information if needed, just let me know what info. Thanks! Diogo Hartmann

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Cant finish call
by przeqpiciel 23 Aug '19

23 Aug '19

Hi all, I am totally new with Kamailio but i have needs to build small cluster of Asterisk. Right now I could succesfully register to asterisk via kamailio and even make a call to hear tt-weasels but Asterisk cant close call My infrastructure diagram is here http://www.asciidraw.com/#Draw6816041626716434245/903175935 This is what probably I need is to know how to tell to Asterisk that send BYE to private IP of Kamailio instead public IP of Kamailio, am I right ?

1 0

TCP: Strange Device Behaviour
by Carsten Bock 23 Aug '19

23 Aug '19

Hi, I have a device here, which does some sort of strange things: I have SIP Message, e.g. with a length of 1100 Bytes and the devices adds "0" to the end to meet the MTU of the network (for a single TCP frame), some sort of TCP padding. The following TCP Frame contains only the next message. Looking at the trace in Wireshark, sngrep or similar, all of them try to re-assemble the next message in the TCP stream, but however don't decode them as SIP due to to first couple of zero's in the message. Looking at Kamailio, it seems not to have any issues with this and decodes/parses the message properly (they show up on Homer properly); however it also forwards the message INCLUDING the first couple of zero's even on UDP. To make it complitely irritating, the phone itself would ignore such packets, if we would forward them to the phone with starting zeros. Has anyone seen anything like this and is it standards-compliant? Any proper and easy way to get rid of the starting "zero's" from a message before forwarding these messages? I could always "hack" something into src/core/tcp_read.c, but I wonder if there is already a proper way? Any ideas anyone? Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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About STIR/SHAKEN - Caller Identity
by Daniel-Constantin Mierla 23 Aug '19

23 Aug '19

Hello, at couple of events I participated during the past few months, I was asked about support of STIR/SHAKEN (caller identity authentication/verification), which is a hot topic these days at least in USA, aiming to combat "fraudulent" robo-calling. Therefore I thought of share some details with everyone in the community about the state in Kamailio, writing to both devs and users, the information being relevant for everyone. We already have the (related) module named auth_identity, available since 2008 (iirc): - https://www.kamailio.org/docs/modules/stable/modules/auth_identity.html But it implements the previous iteration of the specs for caller identity, respectively RFC 4474: - https://tools.ietf.org/html/rfc4474 However, that RFC is obsoleted by 8224 (the latest core specs for STIR/SHAKEN): - https://tools.ietf.org/html/rfc8224 Then, there are also RFCs 8225 and 8226 to add to the core specs. Should anyone be interested to implement STIR/SHAKEN specs in a modules, I would suggest to start from auth_identity -- might not be much work to update it to become conform with latest specs (a new module can be created, of course, even when starting from auth_identity). However, these specs are about signing the SIP request (the INVITE) with special PKI certificate. It can be done easily with embedded scripts such as Lua or Python (inline execution in native kamailio.cfg or using kemi scripts). At Kamailio World 2019, one of the participants I discussed with told me they already implemented using Lua. That's it for a starting point, if anyone wants to discuss more, just reply to sr-users and add your comments or ask the questions. If someone wants to go ahead and work on a C module, announce yourself to avoid duplicate work of others, and use sr-dev if you need assistance on module development. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Unable to start kamailio_pcscf.service
by Jack R 23 Aug '19

23 Aug '19

1. Followed steps from Step # 3 to Step # 16 : https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… 2. When I run last command >systemctl start kamailio_icscf kamailio_pcscf kamailio_scscfthe root@ubuntu18-04:/etc/init.d# *systemctl start kamailio_icscf kamailio_pcscf kamailio_scscf* Job for kamailio_pcscf.service failed because the control process exited with error code. See "systemctl status kamailio_pcscf.service" and "journalctl -xe" for details. 3. All services start but not kamailio_pcscf: root@ubuntu18-04:/etc/init.d# *systemctl status kamailio_pcscf.service* ● kamailio_pcscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_pcscf; generated) Active: failed (Result: exit-code) since Mon 2019-08-19 09:05:30 UTC; 45min ago Docs: man:systemd-sysv-generator(8) Process: 52271 ExecStart=/etc/init.d/kamailio_pcscf start (code=exited, status=1/FAILURE) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: ERROR: bad config file (1 errors) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: * Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Control process exited, code=exited status=1 Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Failed with result 'exit-code'. Aug 19 09:05:30 ubuntu18-04 systemd[1]: Failed to start LSB: Start the Kamailio SIP proxy server. lines 1-16/16 (END) Please advise. Thanks and Regards.

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basic Kamailio frontend to Asterisk
by Travis Ryan 22 Aug '19

22 Aug '19

All, I would like to use Kamailio as a frontend/proxy for an asterisk machine/cluster. I'm very well versed in Asterisk, but VERY noob to Kamailio. I've watched so many Kamailio presentations and feel like I've scoured the web for any article that could help me. There is alot out there, but I'm not finding my exact situation. I believe I understand the dispatcher stuff that I'll need down the road, but I'm having issues understanding the setup of my SIP trunk and Kamailio. I used UAC to setup the trunk connection in Kamailio and it sort of worked. I'm not sure what I do for Asterisk. I'd suspect that in this situation Kamailio would handle the registration and trunk connection for me. What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need to still register the trunk from each Asterisk box "thru" the Kamailio proxy, etc? Also, I'm merely accepting outside calls and then validating the caller and bridging them back out to the PSTN, so I don't have any local SIP clients, etc., so no need to register the sip devices, etc. Thanks! Travis

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Docker image for 5.2.4
by Colin, Andrew 22 Aug '19

22 Aug '19

Hey All Who updates the images on https://hub.docker.com/r/kamailio/kamailio/tags i dont see the latest image kamailio:5.2.4-stretch --

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{param.count,delimiter} issue
by Juha Heinanen 20 Aug '19

20 Aug '19

Looks like there is bug in {param.count,delimiter} transformation: $var(test) = "a=1;b=2"; xinfo("param count = $(var(test){param.count,;})\n"); $var(test) = "a=1,b=2"; xinfo("param count = $(var(test){param.count,,})\n"); syslog: Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 2 Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 1 -- Juha

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sr-users August 2019