sr-users February 2020

sr-users@lists.kamailio.org

57 participants
73 discussions

Same Via Tag for INVITE and ACK on S/L loadbalancer
by Sebastian Damm 11 Feb '20

11 Feb '20

Hi, I stumbled upon an interop problem with a carrier. We have the following scenario: Gateway --> Loadbalancer --> Carrier The loadbalancer generates a Via header for each request. But since it is stateless, the Via tag is generated for each request. As a consequence, the Via tag in the ACK differs from the one in the INVITE. And one carrier doesn't handle those ACKs if the Via tag differs. Is there a way to force the creation of a "deterministic" Via branch tag? For example, building it from a hash over call-id and from-tag or something like that? Thanks in advance Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH

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Different REGISTER expires based on User-Agent
by mayamatakeshi 11 Feb '20

11 Feb '20

Hi, is it possible to reply with a different expires for a REGISTER depending on some condition like the User-Agent? I am thinking maybe I can use: $ulc(profile=>expires) = 600 (but my kamailio dev machine is out of service and I cannot test this right now).

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Kamailio v5.1.10 Released
by Daniel-Constantin Mierla 10 Feb '20

10 Feb '20

Hello, Kamailio SIP Server v5.1.10 stable release is out. This is a maintenance release of the old stable branch, 5.1, that includes fixes since the release of v5.1.9. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.10, to v5.2.x or 5.3.x series. For more details about version 5.1.10 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/02/kamailio-v5-1-10-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is an old stable branch, v5.1.10 being the last planned release out of it, to mark the end of official maintenance. The latest stable branches are now 5.2 and 5.3, at this time with v5.3.2 being the latest stable release. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020 -- www.kamailioworld.com

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Planning the release of v5.1.10
by Daniel-Constantin Mierla 10 Feb '20

10 Feb '20

Hello, I am considering to release Kamailio v5.1.10 on Thursday, Jan 30, 2020, to mark the end of official packaging from branch 5.1. At this moment the last two stable branches are 5.2 and 5.3. If anyone is aware of some commits for fixes that should be backported to branch 5.1, write to sr-dev. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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asterisk kamailio mobile network
by Gintaras Kazakevicius 07 Feb '20

07 Feb '20

Hello, need help. Call from mobile to asterisk: Mobile network provider <trunk> Kamailio <-> Asterisk --- working Call from asterisk to mobile Asterisk <-> Kamailio <trunk> Mobile network provider --- not working How to configure? Maybe you have working example of this solution?

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DIAMETER (server & avp helper)
by Steve Bucklin 07 Feb '20

07 Feb '20

Anyone familiar with: ims_diameter_server cdp cdp_avp I have set Kamailio up to connect with SMSC using Diameter. Initially just want to capture requests in JSON and examine. My trace tells me that I am sending a good message for DIRECT_DEBITING (436). I know the ocs module does not support this - only trying to "take apart" the message. When I get the message, I experience the following: Feb 7 13:07:58 ip-172-31-45-179 /sbin/kamailio[18048]: ERROR: cdp [diameter_avp.c:645]: AAAUngroupAVPS(): hss3g_ungroup_avps: source buffer to short!! Cannot read the whole AVP header! Feb 7 13:07:58 ip-172-31-45-179 /sbin/kamailio[18048]: CRITICAL: cdp [diameter_avp.c:702]: AAAUngroupAVPS(): AVP:<> I suspect this is due to vendor, but not experienced here. The docs seem to suggest the 3gpp is supported, and the diameter xml is set for 0 & 10415? Steve

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Broken link on https://www.kamailio.org/wiki/cookbooks/devel/selects
by Björn Bylander 07 Feb '20

07 Feb '20

Hi, In the section "Complete Select List" on X there's a link to http://sip-router.org/docbook/sip-router/branch/master/select_list/select_l… which doesn't exist anymore. Can anyone tell me where the current list is located? Thanks in advance, Björn Bylander

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Best practice to figure out one way rtp delay
by Karsten Horsmann 06 Feb '20

06 Feb '20

Hi List, I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing. So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios. For testing purposes I setup an freeswitch with beep and then echo application to the caller. My call flow are like this Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together. This generates for the caller an beep and echo. The interesting thing is now, that Call 1 gets an hearable delay of 1 second. But only in the rtp steam from me to the caller. The second calls seems equal of timing. Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no. Now my question to you guys. How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc). Only capturing on one place don't did the trick for me. Thanks for your hints Cheers Karsten

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Binary Websocket messages
by Andre Pires dos Santos 06 Feb '20

06 Feb '20

Hello guys, we are new to kamailio so sorry basic or wrong points. We started using kamailio 5.3.2 and rtpengine as our sip to webrtc solution and SIP.js as webrtc client. We noticed that most of the times webrtc client receives UTF-8 messages. But sometimes webrtc client receives binary messages. When we get binary messages SIP.js cannot parse the message. I couldn't find how to force kamailio to send only UTF-8 messages. Is it possible? We have other issues but I think it will be better for the list to break other points in different messages. Thanks! Santos

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IMS RX AAR issues
by Pafel 06 Feb '20

06 Feb '20

Hello, I am using kamailio 5.3 + rtpengine module + sipwise rtpengine in a IMS test environment. I have and issue with ims_qos module and Rx AAR command. PCSCF is configured with #!define WITH_RX ##!define WITH_RX_REG #!define WITH_RX_CALL I have 2 test phones connected. Both of them use ipv6. PCSCF has ipv4 and ipv6 address. All other nodes - SCSCF, ICSCF use ipv4. Simple scenario UE1 calls UE2. UE2 is replying with 183 session progress and then RX AAR is sent by PCSCF to PCRF/PGW. The problem is that AVPs Media-Sub-Component Flow-Description include ipv4 address of the proxy and ipv6 address of the UE. Example: permit out 17 from 192.168.1.1 34998 to fd18::1:2dca:3303:1286:257d 50010 permit in 17 from fd18::1:2dca:3303:1286:257d 50010 to 192.168.1.1 34998 But this is not acceptable for the PGW (different ip versions). The SDP between UE and PCSCF is rewritten by rtpengine/sipwise. Looking at the ims_qos module I found that the address of the proxy is taken from the session original request SDP media ip/port. Having in mind that rtpengine/sipwise proxy is rewritting it is there any way I can set the correct ipv6 address and port of the proxy? I've tried to get the rewritten media address from SDP (in logs) but without success: if(sdp_get_line_startswith("$avp(cline)", "c=")) { xlog("L_INFO", "SDP media ip: $(avp(cline){s.select,2, })\n"); } if(sdp_get_line_startswith("$avp(mline)", "m=")) { xlog("L_INFO", "SDP media port: $(avp(mline){s.select,1, })\n"); } I am always getting the ipv4 media address and port of the proxy. Any idea how I can set and use the rewritten by the rtpengine/sipwise media ip and proxy for the RX AAR? Thanks in advance, Pavel Siderov

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sr-users February 2020