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TLS traffic visualization HOWTO
by Giovanni Maruzzelli 04 Feb '19

04 Feb '19

# How to TRACE and visualize TLS and non-TLS SIP traffic in real time (thanks to Homer's Lorenzo Mangani for pointing me toward Frida) apt-get install python-pip pip install frida pip install hexdump wget https://raw.githubusercontent.com/google/ssl_logger/master/ssl_logger.py #first ssh terminal # create fifo pipe, then will send the content from fifo pipe to an sngrep without gui, which will be reading pcap from stdin, and sending eep packets to the other sngrep (third terminal) mkfifo /tmp/pipe cat /tmp/pipe | sngrep -N -q -H udp:127.0.0.1:5077 -I - #second ssh terminal # writes as pcap to fifo pipe what freeswitch writes and reads from ssl lib python ssl_logger_giova.py -pcap /tmp/pipe freeswitch #third ssh terminal # sngrep that receives packets from both the Ethernet device, and the eep packets sent by the other sngrep (eg, the tls packets ssl_logger grabs from freeswitch's ssl lib) sngrep -L udp:127.0.0.1:5077 (you may want to edit ssl_logger.py and change 228 to be 101 - LINKTYPE_IPV4 to be LINKTYPE_RAW ) -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Planning Fosdem 2019
by Daniel-Constantin Mierla 02 Feb '19

02 Feb '19

Hello, Fosdem 2019 is approaching, so I am writing to see if any of you plans to go to the event. If there is interest, we can try to organize again a dinner on Saturday evening, a tradition for our project at the past 10 editions or even more. Henning Westerholt will give a presentation about Kamailio in the RTC Devroom, I will be around as well. I know few more developers that plan to go to the event, so let's see who else from the community wants to join us. As usual, at Fosdem will be developers from other VoIP projects, like Asterisk, Janus, CGRates, Homer, Jitsi, ... At the past editions we typically had two "kamailio" events: 1) an "ad-hoc" developers meeting in the cantina (or other available room around) to discuss about short term plans for Kamailio -- time and place being decided as we meet there between us (expected in the afternoon of Saturday or during Sunday). 2) a dinner at a place nearby, with other VoIP folks joining us Reply if you plan to go to Fosdem and say if you want to join for a dinner. Just be aware that you have to pay for your food and drinks at the dinner, unless we are going to be surprised again by a generous sponsor that covers partially or completely to dinner. If you need more details about Fosdem, the website is: - https://fosdem.org Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

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kamailio + asterisk + webrtc
by arish haque 01 Feb '19

01 Feb '19

Hi Daniel, Within a network above tech stack is working fine for VOIP calls. But when i am trying from outside there is no sign of RTP. Cmd for rtp - /sbin/rtpengine -p /var/run/rtpengine.pid --interface=192.168.1.249\!106.51.78.78 --listen-ng 127.0.0.1:60000 -m 50000 -M 55000 --log-level 6 --log-facility local1 106.51.78.78 - public ip 192.168.1.249 - priv ip (kamailio + RTPEngine) Kamailio config when call is b/w WebRTC --> WebRTC rtpengine_manage("trust-address replace-origin replace-session-connection direction=internal direction=external"); Here is SDP, captured on the client sitting outside. v=0 o=root 1133801452 1133801452 IN IP4 106.51.78.78 s=Asterisk PBX 16.0.0 c=IN IP4 106.51.78.78 t=0 0 m=audio 50196 UDP/TLS/RTP/SAVPF 0 8 111 9 126 a=maxptime:60 a=ice-ufrag:5e13a1292fc0b7163d49328b7516f763 a=ice-pwd:4fa5e003195492fa165020c81eff9346 a=candidate:Hc0a801ac 1 UDP 2130706431 192.168.1.172 46810 typ host a=candidate:Hc0a801ac 2 UDP 2130706430 192.168.1.172 46811 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 AA:5A:51:BD:4C:53:65:E4:2B:EC:EB:BF:A9:07:DD:60:E3:46:D8:26:6D:04:C8:21:8B:B9:81:37:3A:EB:55:C0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=sendrecv a=rtcp:50196 a=rtcp-mux a=ptime:20 a=candidate:T7Tqd3oKM4NvATH3 1 UDP 2097152255 106.51.78.78 50196 typ host Looking forward for help from you guys. Thanks & Regards, Arish Haque

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dispatcher alg - 8
by Andrew Chen 31 Jan '19

31 Jan '19

Hey all, I need some more clarification on dispatcher alg 8: “8” - use first destination (good for failover). What is meant by "first destination"? How is that defined if I have more than one carrier. More specifically in our case for some PSTN locations we have this setup: primary carrier - 1 sip trunk backup carrier - 4 sip trunks or primary carrier - 4 sip trunks backup carrier - 1 sip trunk the one that have 4 sip trunks is it going to round-robin through it? this is setup on our old 4.1.5 release. thanks -- Andy Chen achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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IMS early media
by Tsvetan Filev 31 Jan '19

31 Jan '19

Hi all. I want to use SIP early media to play music to the caller in kamailio IMS installation like this: http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html I looked a little bit but didn't find ready solution. The information is vague on this topic. Should this be done through a module or application server ? May I need to handle ringing in onreply_route and send OK with SDP to the caller in SCSCF ? Regards.

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Kamailio/dispatcher routing with SRV record
by Jesse Strahn 31 Jan '19

31 Jan '19

Is it possible to route to the back-end SIP servers via DNS SRV record rather than specifying multiple IP addresses in the dispatcher list on the Kamailio server? Running in AWS and looking to be able to dynamically update my list of available sip servers if a server goes offline or if a server comes online. Have Cloudwatch/Lambda setup to manipulate the SRV record with the IP address additions or subtractions but when I try to put the SRV record as the destination in dispatcher, it fails.

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Kamailio v5.1.7 Released
by Daniel-Constantin Mierla 31 Jan '19

31 Jan '19

Hello, Kamailio SIP Server v5.1.7 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.6. 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 v5.1.7 or to v5.2.x series. For more details about version 5.1.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/01/kamailio-v5-1-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.1 being released out of it. 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 - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

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Planing the release of Kamailio v5.1.7
by Daniel-Constantin Mierla 31 Jan '19

31 Jan '19

Hello, I am considering to release Kamailio v5.1.7 later this week, likely on Thursday, January 31, 2019. As usual, here come the notice to report any issue that you are aware of and not yet registered on the bug tracker in order to give them a chance to be analyzed and eventually fixed before the release. Also, take a look at the branch 5.1 in git and check if there are any missing commits from those you expected to be backported. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com

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Altering received parameter in Contact
by David Cunningham 30 Jan '19

30 Jan '19

Hello, I'm hoping someone can give advice on altering the received parameter in the Contact header on a reply. The situation is that a customer is REGISTERing using TLS, and our Kamailio replies with a 200 OK that includes a Contact header like this: Contact: <sip:123456789@xx.xx.102.10 :5061;transport=tls>;expires=3600;received="sip:xx.xx.102.10:36051;transport=TLS" The customer's device apparently had an issue with this, and the device manufacturer advises that they need the ";transport=TLS" on the received parameter removed. This sounds like a strange requirement, but it's what we've been told. Does anyone know how to alter this in a Kamailio configuration? Our system uses www_authorize to authenticate and is running Kamailio 4.2.1. Thanks in advance for any help! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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Kamailio TCP/TLS Connection getting failed
by arish haque 30 Jan '19

30 Jan '19

Hi All, I am running kamailio 5.2.x on CentOS 7. Call is happening but i am getting the below error continuously in kamailio logs. WARNING: websocket [ws_frame.c:228]: encode_and_send_ws_frame(): TCP/TLS connection get failed Jan 30 17:57:01 localhost /usr/local/sbin/kamailio[22097]: ERROR: websocket [ws_frame.c:765]: ping_pong(): sending keepalive Any clue of why this is happening? Thanks, Arish Haque

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