- sr-users - kamailio.org

Re: [SR-Users] RTPengine with rejected RE-INVITE when port is changing
by Daniel Tryba 05 Dec '19

05 Dec '19

On Thu, Dec 05, 2019 at 09:37:51AM +0000, Laurent Schweizer wrote: > Hello, > > I already see old post about this : > https://opensips.org/pipermail/users/2014-November/030451.html > > but I???m interested to know if now they is a solution ???? > > so the issue is a RE-INVITE rejected (415 Unsupported Media or 488 ) and the RTP port is changing. > Is that case the old RTP relay (I???m using RTPengine) is destroyed after 30sec. > Any way to restore them ? I'm confused: - opensips != kamailio - rtpproxy != rtpengine But RFC 3261 states (page 16): During the session, either Alice or Bob may decide to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re- INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. The other party sends a 200 (OK) to accept the change. The requestor responds to the 200 (OK) with an ACK. If the other party does not accept the change, he sends an error response such as 488 (Not Acceptable Here), which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. Full details on session modification are in Section 14. So if this happens with kamailio/rtpengine that is a bug in (I presume) rtpengine.

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siputils: how does sip_p_charging_vector generate the header
by George Diamantopoulos 05 Dec '19

05 Dec '19

Hello, I've been looking at the sip_p_charging_vector() function of siputils. From the documentation it is not evident how the PCV header is generated. Are the exported $pcv pseudovariables writable, so that the function canl use those when generating the header? Thanks! BR, George

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anycast + dmq_usrloc
by Nuno Miguel Reis 05 Dec '19

05 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replication the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: *anycast node that actually gets the SIP REGISTER:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *anycast node that gets the USRLOC update via KDMQ:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *the actual KDMQ message:* *KDMQ sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060> SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch=z9hG4bK6034.290e4186000000000000000000000000.0 To: <sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060>> From: <sip:usrloc@10.100.100.22:5060 <http://sip:usrloc@10.100.100.22:5060>>;tag=3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 <0cfde4d32a31addc-8875(a)127.0.0.1> Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip:2.2.2.2:31171 <http://2.2.2.2:31171>","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0}* -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra mobile: +351 918519145 -- get my PGP key @ http://pgp.dei.uc.pt:11371/pks/lookup?op=get&search=0x750296EECEAD2B6D

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anycast + dmq_usrloc
by Nuno Miguel Reis 04 Dec '19

04 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replicating the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: anycast node that actually gets the SIP REGISTER: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 [http://1.1.1.1:5060/] Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } anycast node that gets the USRLOC update via KDMQ: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } the actual KDMQ message: KDMQ sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch= z9hG4bK6034.290e4186000000000000000000000000.0 To: < sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] > From: < sip:usrloc@10.100.100.22:5060 [http://sip:usrloc@10.100.100.22:5060/] >;tag= 3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 [mailto:0cfde4d32a31addc-8875@127.0.0.1] Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] ","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0} Looking forward to hear from you. BR, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

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Vendor Specific RBT Call Flow
by Hamid Hashmi 04 Dec '19

04 Dec '19

Dear Users Can the following call flow be implemented, if yes, can you give some suggestions with modules information? [cid:d15f60e7-73ec-4cd8-8954-92f472a94ef7] Regards Hamid R. Hashmi

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siremis help and usage index?
by PICCORO McKAY Lenz 04 Dec '19

04 Dec '19

it's there some help manual usage for siremis? the only KB source i have are asipto.. and only are install manuals!! Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

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Re: [SR-Users] kamailio IMS supporting VOLTE
by Shubhendu Das 04 Dec '19

04 Dec '19

Hi , Does Kamailio has its own HSS supporting CX Interface ? BR, Shubhendu From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Mojtaba Sent: Saturday, November 23, 2019 4:37 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] kamailio IMS supporting VOLTE Hello, Running the IMS network for supporting VoLTE with Kamailio module and FHOSS is challenging. THe Kamailio works great in IMS network but You have to optimise your own database. Using FHOSS without any support or development is very risky. In general speaking, running VoLTE have three main section and the IMS network is one of them. You might have some challenges in EPC and RAN access. WIth Best Regards. Mojtaba On Thu, Nov 21, 2019 at 3:23 PM Shubhendu Das <shubhendu.das(a)azcom.in> wrote: Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- --Mojtaba Esfandiari.S -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 1

Fwd: app-ads.txt monetization proposal
by Joel Serrano 04 Dec '19

04 Dec '19

Not sure if this is BS or worth exploring, it's not my expertise area so I'm forwarding to you all just in case ---------- Forwarded message --------- From: Lorenzo from Instal <lorenzo.tacchio(a)instal.com> Date: Tue, Dec 3, 2019 at 8:56 AM Subject: app-ads.txt monetization proposal To: <joel(a)textplus.com> app-ads.txt monetization proposal Hello , I am Lorenzo, Senior Account Manager at Instal; we are currently buying your app's traffic via DSPs. We thought would be beneficial to work directly with you, *monetizing your inventory with no intermediate levels at all*. The key of our success is the *simple technical integration*. No SDK is required, we will add our ads.txt to your page and your app would be included in our media buying plan with budgets of *more than $1M per month*. Would you be interested to discover more in detail this opportunity? Please share your Skype ID or add me, *live:loresseintes*, to connect further! Kind Regards, Lorenzo Instal Srl, Via Ippolito Pindemonte, Florence, Italy Manage preferences <https://hs-3923849.s.hubspotemail.net/hs/manage-preferences/unsubscribe?d=V…>

1 1

secfilter usage
by Aymeric Moizard 03 Dec '19

03 Dec '19

Hi all, I'm testing and moving my kamailio script to use the newer secfilter module. In the past, I was replying "200 Ok" to "friendly scanners"... With the newer secfilter module, it looks like I can't send a reply with "sl_send_reply("200", "OK")" secf_check_ua(); if ($? == -2) { sl_send_reply("200", "OK"); exit; } I have read the code of the secfilter, but I was not able to see any specific code to silently discard the SIP request. I can see in the documentation about "secf_check_sqli_all", that the SIP message is supposed to be "dropped". I can see "w_check_sqli_all" returns 0 on detection and w_check_ua returns -2 upon detection. Are the message discarded because a negative value was returned? Would it be doable, using the secfilter, to still reply 200 Ok? Regards Aymeric -- Antisip - http://www.antisip.com

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kamailio IMS supporting VOLTE
by Shubhendu Das 03 Dec '19

03 Dec '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 2

Contact header requirement
by David Villasmil 02 Dec '19

02 Dec '19

Hello guys, I'm connecting to a provider who requires the contact contains the proxy's IP... I know this is weird and totally off-rfc... I setup the topoh module and used modparam("topoh", "mask_ip", "[PROXY_IP]:5060") but this makes the contact: Contact: <sip:[PROXY_IP]:5060;line=sr-Z213nshDqt98-2qiIoT8GFf. y8PWype.SRy0nRo3nCT*> I know this is how it works, but they don't like the additional ";line=...." Has anyone faced this kind of thing? is there any other way to set this to something like: Contact: <sip:[PROXY_IP]:5060> that won't mess everything? Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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$null + 0 is not int
by Juha Heinanen 02 Dec '19

02 Dec '19

Why is it that $null + 0 is 0 but not int? What is it? -- Juha $avp(test) = $null + 0; if (is_int("$avp(test)")) INFO(test $avp(test) IS int); else INFO(test $avp(test) IS NOT int); Dec 1 02:46:01 salmon /usr/bin/sip-proxy[10138]: INFO: test 0 IS NOT int

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Kamailio World 2020 - Call For Presentations
by Daniel-Constantin Mierla 02 Dec '19

02 Dec '19

Hello, I would like to announce that Call for Presentations at Kamailio World 2020 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k08/call-for-speakers/ The 8th edition of the event takes place again in Berlin, Germany, during April 27-29, 2020. Expect over 150 participants, developers and community members as well as representatives of other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Possible conflict between t_newtran and setflag for ACC
by Patrick Wakano 01 Dec '19

01 Dec '19

Hello list, Hope you all doing well! I am using the ACC module and using the setflag() function as done in several examples. It works fine. However, I've added the t_newtran() function almost in the begging of the INVITE handler to help the retransmission detection and after that I noticed the ACC was not saving anything in DB. So after debugging I discovered that if I call the t_newtran() before setting the ACC flags, the module will not save the calls in DB, but if I call it after setting the ACC flags, it works.... So my question is, is this a bug or it is a expected side effect so when one is using t_newtran you must be careful and set all your transaction flags before? (ACC are the only transaction flags I am using so can't tell if other modules have the same problem) This is happening in Kamailio 5.2.2. Thank you! Kind regards, Patrick Wakano

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tcp limit, too many open files
by David Villasmil 01 Dec '19

01 Dec '19

Hello guys, I'm facing an issue where, if i start kamailio as root, all works properly, but if i start it as kamailio, i get hundreds of: handle_new_connect(): error while accepting connection(24): Too many open files My limits is like: kamailio soft core unlimited kamailio soft data unlimited kamailio soft fsize unlimited kamailio soft memlock unlimited kamailio soft nofile unlimited kamailio soft rss unlimited kamailio hard stack 240 kamailio soft cpu unlimited kamailio soft nproc unlimited kamailio soft as unlimited kamailio soft priority -11 kamailio soft locks unlimited kamailio soft sigpending unlimited kamailio soft msgqueue unlimited kamailio soft nice -11 kamailio soft nofile 65536 kamailio hard nofile 65536 root soft core unlimited root soft data unlimited root soft fsize unlimited root soft memlock unlimited root soft nofile unlimited root soft rss unlimited root hard stack 240 root soft cpu unlimited root soft nproc unlimited root soft as unlimited root soft priority -11 root soft locks unlimited root soft sigpending unlimited root soft msgqueue unlimited root soft nice -11 root soft nofile 65536 root hard nofile 65536 I can't figure out why this is happening... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Kamailio as SBC for Microsoft Teams
by Henning Westerholt 29 Nov '19

29 Nov '19

Hello, I have created a "How-To" blog post on using Kamailio as "session border controller" for Microsoft Teams Direct Routing: https://skalatan.de/de/blog/kamailio-sbc-teams You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

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Adding "q" value for the main branch
by Maxim M 28 Nov '19

28 Nov '19

Maxim Malygin <maxim.malygin(a)gmail.com> 17:08 (5 минут назад) кому: sr-users Hello, I need to form list of Contact headers with "q" parameters for "302 Moved Temporary". I do something like this: ... # $xavp(contacts) - contains list of contact URIs (uri) and Q values (q) $var(i) = 0; $var(num) = $cnt($xavp(contacts)); while($var(i) < $var(num)) { if($var(i) == 0) { # How to add q value to this branch? $ru = $xavp(contacts[$var(i)]=>uri); } else { append_branch("$xavp(contacts[$var(i)]=>uri)", "$xavp(contacts[$var(i)]=>q)"); } } send_reply("302", "Moved Temporarily"); ... Is it possible to set "q" value for the main branch? For additional branches I do it via append_branch("uri", q_value) but I didn't find any way to do it for the main branch pointed via $ru/$du. The pseudo-variable "$branch" also doesn't allow me to access the main branch. Thanks, Maxim

1 0

Get totally rid of H.264 SDP
by Kjeld Flarup 28 Nov '19

28 Nov '19

Hi I have a door telephone, with GSM fallback. Since telephone operators are going away from SS7, more and more traffic goes SIP. Thus we now have some numbers where we can see that doing an invite with H.264 causes them to fail. I have tried to remove coded 96 with sdp_remove_codecs_by_id, but it still leaves most of the video SDP in the invite. SDP received by kamailio Media Description, name and address (m): video 6100 RTP/AVP 96 Media Attribute (a): rtcp:6101 IN IP4 192.168.237.239 Media Attribute (a): rtpmap:96 H264/90000 Media Attribute (a): fmtp:96 profile-level-id=42000d;packetization-mode=1 Media Attribute (a): sendrecv After sdp_remove_codecs_by_id I send this. But without "96" Media Description, name and address (m): video 6534 RTP/AVP Media Attribute (a): sendrecv Media Attribute (a): rtcp:6535 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): candidate:r9KO0wNPMTGktklS 1 UDP 2130706431 192.168.237.15 6534 typ host Media Attribute (a): candidate:r9KO0wNPMTGktklS 2 UDP 2130706430 192.168.237.15 6535 typ host My problem is that the door telephone does not do an ACK when it gets the 200 OK My suspicion is that it does not understand the video response Media Description, name and address (m): video 0 RTP/AVP 0 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): sendrecv Is there a way to remove the whole video part and not just the coded id? It is not an option to disable video in the door telephone, as we need the video. It is just in the fallback is must be removed. -- --------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

1 0

Kamailio does not detect message
by Adesh Pandey 28 Nov '19

28 Nov '19

Hi, I have below scenario: User registers on Kamailio server, calls to a number which is forwarded to asterisk through dispatcher, asterisk server answers and sends a text message to the caller using sendText but first message is delivered to client with message "501 Not Implemented" but in the subsequent calls asterisk shows that the text was delivered but kamailio did not detect the message. Guys, any help would be appreciated, I am using the below mentioned config. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Kamailio version: 5.2

1 0

Kamailio developers meeting - follow up remarks
by Daniel-Constantin Mierla 27 Nov '19

27 Nov '19

Hello, last week we had the 2nd annual Kamailio Developers Meeting hosted by Sipgate in Dusseldorf, Germany: * https://www.kamailio.org/w/developers-meeting/ 16 people were at the event in various roles. Giacomo Vacca published on his blog a good summary of what happened there: * https://www.giacomovacca.com/2019/11/my-notes-on-kamailio-developer-meeting… This year we had a lot of discussions, as well as work done on multiple planes, not only Kamailio code. So I am trying to list here some of the conclusions for future development, the technical aspects of the meeting, so everyone is aware and can provide feedback. 1) Effective work was done on: * kamailio code * kamailio rpm packaging * kamailio tools (kamctl) * kamailio release process * kamailio project keys (to be used to sign the packages) 2) Documentation 2.a) Wiki * it was somehow a rough consensus to move the wiki content to github, along with changing the format from dokuwiki markdown to the standard/github markdown. This should enable people to make pull requests so developers or community members can review and aprove new content. It also makes it easier to contribute using existing github account, now the kamailio.org/wiki is requiring to make a dedicated account, which many prefer not to do it. * the presentation can be done either by using mkdocs to generate html files hosted on kamailio.org or using the github provided wiki portal. 2.b) Docs for variables and transformations * there was a proposal to move them in the documentation the modules that export them, there are pros and cons, needs more discussions. Now they are in the wiki, so this probably has to resume after deciding on 2.a). 3) Kamailio Modules 3.a) replication (dmq) - several participants discussed about negotiations between nodes to take active role on some cases (e.g., active dialogs) 3.b) api integration - quite some interest in JSON-based API routing, concluding in extending rtjson to cover more use cases 3.c) security - have options to restrict the use of TLS1.3 or newer 4) Kamailio Releases * v5.3.2 was released during the event, allowing to document the process * work to automatize the process is planned, then eventually assing teams for takeing cares of releases from specific branches 5) Kamailio Testing * existing docker-based testing framework should be extended and integrated in CD/CI pipeline 6) Kamailio packages 6.a) rpms * rpm.kamailio.org has been prepared and is expected to take over the opensuse build service for building rpms and hosting them. Expected to provide support for hosting many kamailio versions in the same release series so one can do downgrade to older releases. Also, there is work in progress to provide nightly builds. 6.b) debs * work is planned to offer many kamailio versions in the same release series 7) Kamailio tools * kamctl/kamdbctl should be obsoleted in favor of kamcli, which offers a better framework for input validation and output formatting, as well as better portability, no longer depending on shell interpreter 8) Various discussions * kemi exports from C point of view and how to combine the documentation for modules and their kemi exports * how to make kamaiio friendlier in virtualized environments (ended up in the need of making the use of advertised address a bit more dynamic) * project organizatoric topics - to be approached separately * next events - Fosdem - someone should submit a proposal to present about Kamailio 9) Long term goals We speak here more or less about Kamailio 6.0 ... * change the behaviour of the native config interpreter to be consistent with the other programming languages in terms of handling the response code (change what is now: the evaluation of negative value to false and positive value to true and the hidden return 0 to exit) * make the pool of processes more generic, so they can handle traffic from more sockets (being sip traffic or something else) -- this should make better use of resources, as some sockets might be less busy that others I hope I covered the important topics, if I remember something else, I will reply on this thread. Or maybe other participants can contribute missing topics. Should anyone have comments or suggestions on the above topics, or new ones, let's discuss on sr-users because it impacts the long term use of Kamailio (sr-dev is cc-ed now for notification purposes). Overall, there were 2 very intensive days, however in a friendly and relaxed environment offered by Sipgate. Extremely useful discussions, not only about Kamailio, but about RTC ecosystem. At the evening social network event event, a couple of external people joined, some of them presenting very interesting use cases of Kamailio. Many thanks to all participants that allocated time and resources to come to Dusseldorf, as well as to the companies that covered expenses for participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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Kamailio xavp_contact - Segmentation Fault
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, Kamailio is crashing when i'm trying to set the parameter; modparam("tm|usrloc", "xavp_contact", "ulattrs") That's happening with Kamailio 5.3.1 and 5.2.5 too. Crash is happening when Im register 2 devices with the same extension. This is the core dump: [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /usr/local/etc/ka'. Program terminated with signal 11, Segmentation fault. #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 288 core/xavp.c: No such file or directory. (gdb) bt #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 #1 0x0000000000581869 in xavp_insert (xavp=0x0, idx=0, list=0x0) at core/xavp.c:752 #2 0x00007f05be0ee3a1 in ki_t_next_contacts (msg=0x7f05cb76c218) at t_serial.c:552 #3 0x00007f05be0f09e8 in t_next_contacts (msg=0x7f05cb76c218, key=0x0, value=0x0) at t_serial.c:756 #4 0x0000000000434d41 in do_action (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1067 #5 0x00000000004418e8 in run_actions (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1572 #6 0x0000000000441fa9 in run_actions_safe (h=0x7ffe93c9de90, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1636 #7 0x000000000065734e in rval_get_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, i=0x7ffe93c9cf78, rv=0x7f05cb642570, cache=0x0) at core/rvalue.c:912 #8 0x000000000065b8fe in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9cf78, rve=0x7f05cb642568) at core/rvalue.c:1910 #9 0x000000000065bd51 in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9d42c, rve=0x7f05cb642c98) at core/rvalue.c:1918 #10 0x0000000000434807 in do_action (h=0x7ffe93c9de90, a=0x7f05cb645028, msg=0x7f05cb76c218) at core/action.c:1043 #11 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb637e48, msg=0x7f05cb76c218) at core/action.c:1572 #12 0x0000000000431767 in do_action (h=0x7ffe93c9de90, a=0x7f05cb583858, msg=0x7f05cb76c218) at core/action.c:691 #13 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb574750, msg=0x7f05cb76c218) at core/action.c:1572 #14 0x0000000000442071 in run_top_route (a=0x7f05cb574750, msg=0x7f05cb76c218, c=0x0) at core/action.c:1657 #15 0x00000000005874a9 in receive_msg (buf=0xa6ec00 <buf.6868> "OPTIONS sip:10@192.168.0.231:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.231;branch=z9hG4bKd8a9.1b3abe656532065414a2f35a8916008d.1\r\nVia: SIP/2.0/UDP 192.168.0.131:5060;received=192.168.0.131;branch=z9hG"..., len=694, rcv_info=0x7ffe93c9e4d0) at core/receive.c:341 #16 0x000000000047bf81 in udp_rcv_loop () at core/udp_server.c:541 #17 0x0000000000424f9d in main_loop () at main.c:1669 #18 0x000000000042c688 in main (argc=13, argv=0x7ffe93c9ea18) at main.c:2710 Thank you.

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Kamailio Parrallel Forking
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, I'm trying to use parallel forking if one extension is registered from multiple devices. (using WebSockets) This is the related code from kamailio.cfg: ------------------------------------------------------------------------------------------------------------------------------------ if (!t_load_contacts()) { xlogl("L_WARN", "Error loading contacts for $rU\n"); sl_send_reply("500", "Server Internal Error (Code:$cfg(line))"); exit; } else { xlogl("L_INFO", "Contacts loaded for $rU\n"); } if (!t_next_contacts()) { xlogl("L_INFO", "t_next_contacts - Only one contact found for $rU, calling\n"); } else { xlogl("L_INFO", "t_next_contacts - Multiple contacts found, parallel forking\n"); } route(RELAY); ------------------------------------------------------------------------------------------------------------------------------------ I noticed that on Kamailio 5.2.5 - this code is working fine, and Kamailio is sending an INVITE to both Devices.But, on Kamailio 5.3.1 - that's not working anymore. For some reason, kamailio is sending just one INVITE to one device with wrong URI. The URI is set to the second device instead of the correct one. Was something changed in the t_next_contacts function related to this? Thank you

1 0

Kamailio 5.2.3 segfaulting
by Daniel Greenwald 27 Nov '19

27 Nov '19

I'm seeing this in the log and kamailio is crashing every hour or so: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 33/var/log/kamailio/kamailio.log Any ideas? Should I emergency upgrade to 5.3.1?

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No Audio using RTP Proxy
by Sujit Roy 27 Nov '19

27 Nov '19

Hello I am facing a problem as below. Please suggest for the work around. My call flow is like this. SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy So when the call arrives at Kamalio+RTPProxy, i m getting below in log. Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@ 192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_command_ul_handle: new session on IPv4 port 15920 created, tag 1aa18fc201a68168;1 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTP address with 192.168.100.177:27360 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTCP address with 192.168.100.177:27361 But x.177 is working on signalling mode only ( Not routing Media ) . As a result, i m not getting any voice from IP x.179 What can be done to change the caller's RTP address to x.179 in RTPProxy ? Thanks -- Regards =================== Sujit Roy

2 1

Encode host ip in Contact header for semi-stateless setup
by Daniel Greenwald 27 Nov '19

27 Nov '19

At kamailio world 2017, Jan Lorenz of Pascom explained how they use kamailio in a semi-stateless manner and encode the asterisk's IP into the contact header on the way out so that it can be decoded and used to route properly on the way back without Record-Routing. I would like to create such a setup. I'm looking for pointers if there is a particular module/method that would be ideal to to use for this. Any other gotcha's to consider in such a setup. My plan is to keep the transaction on the same kamailio by not altering the VIA but allow the subsequent in-dialog transactions to potentially hit another kamailio by changing the contact to a shared SRV record (and encoding the freeswitch box's IP into it for later retrieval.) Thanks in advance. Daniel Greenwald

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