June 2021 - sr-users - kamailio.org

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

4 10

Dialog and DMQ on restart
by Björn Klasen 25 Mar '24

25 Mar '24

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

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Is Min-SE check broken?
by harneet singh 31 May '23

31 May '23

Hi All, I am facing an issue in understanding how the min_se should be working in kamailio. As per the SST documentation, it seems like if the min_se is configured as 500, then any value of Session-Expires OR MIN-SE if lower than 500, can be responded to by a 422. However, I strangely see the reverse happening. To investigate further, I looked in to the ki_sst_check_min() code in the master, and these seems like a potential issue. Ref Code: Inside ki_sst_check_min(), there is an if condition like below: if (sst_min_se < MIN(minse, se.interval)) { However, shouldn't it be the other way around? ie if (sst_min_se > MIN(minse, se.interval)) { because we need to send 422 if the received value(in INVITE etc) is smaller than the sst configure min_se value? I also found a different documentation, at https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6… where the check is as below? if (sst_min_se > MIN(minse, se.interval)) { Can someone confirm if this is broken, or my understanding is incorrect? Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 02 May '23

02 May '23

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

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Possibility of mirroring the rpm/deb.kamailio.org repos efficiently via rsync?
by Malte Schmidt 23 Jun '22

23 Jun '22

Hello, is there an rsync endpoint available or is there a possiblity of setting this up? Creating a mirror via HTTP is a rather quick and dirty solution and while the deb-repo can be mirrored using debmirror, the rpm repo is hard to sync to a non-CentOS-based machine due to missing dependencies such as yum and reposync in latest Debian-based systems. Would be great to get some input in regards to this topic. Cheers

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Fwd: Possible memory leak on 5.5.x (new)?
by George Diamantopoulos 19 Jan '22

19 Jan '22

Forwarding my reply to the list, using gmail's reply button set Henning as the sole recipient :-\ ---------- Forwarded message --------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: Sat, 26 Jun 2021 at 02:25 Subject: Re: [SR-Users] Possible memory leak on 5.5.x (new)? To: Henning Westerholt <hw(a)skalatan.de> Hello Henning, Thanks for your reply. Here's what has come up after a few hours: shm55: https://pastebin.com/h9JCePmc shm54: https://pastebin.com/Nx5xEEnA It seems to me htable is the culprit? Are you seeing anything different? 54 has been running for 77020 seconds, 55 for 28521 (significantly less). I'm going to turn it off until we figure something out... BR, George On Fri, 25 Jun 2021 at 18:17, Henning Westerholt <hw(a)skalatan.de> wrote: > Hello, > > > > Good observation. Please run the memory statistics CLI commands to get > more hints about the module that might cause it (as per below link). Then > please report more details. If you can point to a particular module, you > can also open an issue on our tracker. > > > > https://www.kamailio.org/wiki/tutorials/troubleshooting/memory > > > > Cheers, > > > > Henning > > > > *From:* sr-users <sr-users-bounces(a)lists.kamailio.org> *On Behalf Of *George > Diamantopoulos > *Sent:* Friday, June 25, 2021 4:53 PM > *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > *Subject:* [SR-Users] Possible memory leak on 5.5.x (new)? > > > > Hello all, > > > > I'm still investigating the (most likely non-kamailio-related) memory leak > of my previous message to the list, there have been no developments so far. > I'll update if anything changes. > > > > This concerns a new finding which seems to affect kamailio 5.5.x. I have > two kamailio instances receiving the same traffic via round-robin. I > upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared > of any issues arising. I was lucky to do so, because with identical > configuration, 5.5.x seems to run out of SHM very quickly. Here are links > to graphs produced by our monitoring system: > > > > Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png > > New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png > > > > The configuration uses mtree, htable, vars and vns extensively. Has anyone > come across anything similar? Let me know if I can provide any further > information to help disect this. Thanks! > > > > BR, > > George >

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ENV Values in cfg
by H Yavari 06 Dec '21

06 Dec '21

Hi to all, Is it possible to set modules parameters with env values? I know that in routes we can use $env(NAME), but what about for other parameters? for example setting DB address in K8s cluster ENV values. Regards,Hossein

2 1

update max_while_loops in config
by Juha Heinanen 08 Oct '21

08 Oct '21

It is possible to update value of max_while_loops dynamically in config file? For example, there could be a need to increase the max in some while loop, but after the loop is done, decrease it again for other loops. There exists an rpc command to do that, but I haven't found how to do it in config file. I tried @cfg_seti.core.max_while_loops = 200; but got syntax error. -- Juha

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manually append_branch to lookup result
by David Villasmil 07 Sep '21

07 Sep '21

Hello guys, I'm trying to dynamically add a branch after doing lookup. The user is found, but in some cases I need to add a branch and do parallel forking. So i'm basically doing: route[LOCATION] { if (!lookup("location")) { .... } if (something) { route(BRANCH_TO_EXTRA); } route(RELAY); } route[BRANCH_TO_EXTRA] { $fs = MY_SOCKET; append_branch("sip:$tU@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port)); return; } For some reason only the branch appended is being used (I have append_branches=1) Ideas? David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [SR-Users] t_drop_replies not working with t_suspend in failure route
by Huber Andreas 03 Sep '21

03 Sep '21

Hi, We’re still using kamailio 4.4 but we’ll be migrating to 5.0 soon. Cool so it will be fixed when we migrate ! Thanks, Andreas From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Federico Cabiddu Sent: vendredi 12 mai 2017 11:56 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] t_drop_replies not working with t_suspend in failure route Hi, which version are you using? A similar case had been reported some months ago and it should be fixed in 5.0. Regards, Federico On Fri, May 12, 2017 at 11:44 AM, Huber Andreas <andreas.huber(a)nagra.com<mailto:andreas.huber@nagra.com>> wrote: Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We’d like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Kamailio v5.4.6 Released
by Daniel-Constantin Mierla 13 Jul '21

13 Jul '21

Hello, Kamailio SIP Server v5.4.6 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.6. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.0 being release out of it. For more details about version 5.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-4-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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 Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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SECSIPID Identity Decode
by Daniel W. Graham 06 Jul '21

06 Jul '21

I am unable to base64url decode the json payload in identity header generated by secsipid. (Using python for test) decoded_payload = url64.decode(‘payload’) UnicodeDecodeError: 'utf-8' codec can't decode byte 0xc2 in position 27: invalid continuation byte Header decodes fine this way but not payload. Is this an issue with the payload encoding? Kamailio 5.5 Daniel W. Graham, CTO CMSInter.net LLC DIRECT (989) 400-4230 INTERNET | TELEPHONE | MANAGED IT

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Kamailio BSD license
by stevecarellccie＠yahoo.com 01 Jul '21

01 Jul '21

Hello, I am trying to use Kamailio in our application which would be deployed in certain environment in addition to a cloud infrastructure. As I understand it, parts of the Kamailio code are covered under the BSD license - are you able to share what those are? I am trying to determine if Kamailio can be used in a customer environment or should it be strictly restricted to cloud environments due to the GPL v2 licenses Thanks,Steve

2 1

install stirshaken module by apt at Debian 10
by Yuriy Nasida 30 Jun '21

30 Jun '21

Hi, I know that it's possible to compile manually but what about apt method to install stirshaken module? I am checking the list of modules using: apt search kamailio But... Nothing looks like stirshaken. Thus, only several modules available and for others we should compile manually ? Please advise Thanks

2 1

Websocket In-dialoge SIP routing failed post network loss due to aliases
by Shahid Hussain 30 Jun '21

30 Jun '21

Hi, Websocket module documentation has a code reference to use aliases for SIP routing. However, aliases will not work in the following setup and situation. 1. Kamailio is configured with active and standby node 2. Ping is implemented from webclient and kamailio responds with pong. 3. Two clients ClinetA and ClientB registered themselves to Kamailio. 4. After SIP negotiation (INVITE-200OK) each client learnt about below aliases. Alias of ClientA: alias=172.27.6.98~58336 Alias of ClientB: alias=172.27.6.98~58342 So normally if ClientB wants to send SIP message to ClientA, SIP URI from ClientB looks like below ACK sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 4. Call is in a connected state. Following is the issue. i. Switchover (or network lost or reboot) at Kamailio happened ii. Due to ping pong both the clients detected network loss individually and re-registered themselves. iii. Aliases of both the clients got changed. New Alias of ClientA: alias=172.27.6.98~ 58346 New Alias of ClientB: alias=172.27.6.98~ 58348 iv. But ClientB doesn’t know that ClientA also re-registered and ClientA’s alias got changed and vice-versa v. Because of this ClientB still sends BYE with Initial alias BYE sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 Would like to know what is the recommended solution for this problem using alias or is it a limitation of using alias? Thanks & Regards, Shahid

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Re: [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy
by Peter Manley 29 Jun '21

29 Jun '21

Thanks Muhammad, That seems to have done the trick. I am now able to bridge WebRTC to UDP. Regards, Peter Manley M S shaheryarkh at gmail.com<mailto:sr-users%40lists.kamailio.org?Subject=Re%3A%20%5BSR-Users%5D%20Guidance%20building%20a%20WebRTC%20to%20Asterisk%20Intermediate%0A%20Proxy&In-Reply-To=%3CCAFZQphyZE_tGB9%3D5tFQBtXAv5twMh%3D9aCNFAvveGf0-L%3Dy_qwQ%40mail.gmail.com%3E> Sat Jun 26 03:57:27 CEST 2021 * Previous message (by thread): [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy<https://lists.kamailio.org/pipermail/sr-users/2021-June/112799.html> * Messages sorted by: [ date ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/date.html#112800> [ thread ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/thread.html#112800> [ subject ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/subject.html#112800> [ author ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/author.html#112800> ________________________________ The problem seems to that the kamailio is trying to route it over UDP port using WSS transport, which will never work. You are creating a transport bridge (UDP <-> WSS i.e. stateless transport <-> stateful transport) as well media bridge (RTP <-> SRTP), so it is going to be tricky. I strongly discourage it since things get messy for sequential requests. But you can try the following, Here are some PATH module parameters and functions, you will need to setup. 1. You will need to enable "use_received" parameter to ensure connection tracking. 2. It seems you may have enabled "received_format", so you make sure you do NAT detection and use "handle_ruri_alias" for initial requests. 3. You must also enabled "enable_r2", but with caution, since both ends (i.e. UAC and UAS) can be using same kamailio (proxy) socket for in/out. 4. Make sure to use "add_path_received" instead of "add_path". 5. If using OUTBOUND module then make sure it is loaded BEFORE the PATH module. Thank you. -- Muhammad Shahzad Shafi Burraq Technologies Tel: +49 176 99 83 10 85 On Sat, Jun 26, 2021 at 12:47 AM Peter Manley <peter.manley at kombea.com<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > Hello, > > > > I'm working on building an RTP Proxy which will allow both regular SIP to > SIP transactions and WebRTC to SIP transactions to an Asterisk Server. > > > > SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > AND > > > > WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > The Kamailio/RTPEngine Proxy is stateless (does not use the tm or > registerer modules), and uses the Path module to replace the Route header > with a Path header. > > > > It works grep for SIP/UDP to SIP/UDP transactions. > > > > HOWEVER... It only partially works for WSS to SIP/UDP transactions. The > REGISTER/401(Unauthorized), REGISTER/200(OK) transaction works well. > > > > When the Asterisk server sends a SIP Options message, I get the following > error: > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: <script>: => OPTIONS < > sip:l4umfumh at 192.0.2.97<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>;transport=ws;alias=209.141.117.212~59751~6 -> > <null> (IP:172.16.0.48:sp) > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/msg_translator.c:2815]: via_builder(): TCP/TLS connection (id: 0) for > WebSocket could not be found > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/msg_translator.c:2003]: build_req_buf_from_sip_req(): could not > create Via header > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/forward.c:550]: forward_request(): building failed > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/receive.c:425]: receive_msg(): error while trying script > > > > It appears that the WebSocket corresponding to the WSS leg cannot be > located when the request is initiated from the Asterisk Server. > > > > Any thoughts on how I can get this to work in the other direction > (Asterisk to Kamailio)? I've been looking at the WebRTC to SIP Client and > Server (https://github.com/havfo/WEBRTC-to-SIP), but that doesn't quite > do what I want it to do (i.e. it appears to act as a registrar server and > doesn't forward REGISTER requests on to another registrar). > > > > > > Thanks and regards, > > > > *Peter H. Manley, Ph.D.* > > Chief Software Architect > > KomBea Corporation > > 3400 North Ashton Blvd, Suite 470 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > Lehi, Utah 84043 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > www.kombea.com<http://www.kombea.com> > <https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> > > > > Office:+1.801.610.5808 > > Mobile +1.801.243.7773 > > Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

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Kamailio + RTPEngine - Destinations Behind NAT
by Edward Romanenco 29 Jun '21

29 Jun '21

Hi everyone, I'm using Kamailio (Ver 5.5.1, Public IP X.X.X.11, Private IP 172.18.0.20) + RTPEngine (Ver 9.4.1.1, Public IP X.X.X.7, Private IP 172.18.0.50) alongside Freeswitch (Private IP 172.18.0.40) in my backend and delivering calls to a mobile application (call flow is Freeswitch -> Kamailio + RTPEngine -> Applicaiton). While calls made to NATted destination generally works fine - from time to time, we've experiencing one-sided audio due to the rtpengine sending the media stream to the remote end's private IP instead of his public one. For instance, in the attached logs - media was sent towards 192.168.1.93 instead of 79.179.1.2, which is the expected destination. Problem appears almost randomly; I can make 20 fine calls in a row for the same destination until experiencing it, and I didn't find any difference as it comes to the SIP invite exchange between the two. Has anyone ever experienced the same behavior and can share some general instructions on how to troubleshoot it? Relevant Kamailio config section is attached. Invite Kamailio->Device INVITE sip:972587102881@79.179.1.2:56308;transport=TLS;ob SIP/2.0 Record-Route: <sip:X.X.X.11:443;transport=tls;r2=on;lr=on;ftag=cvcHy5UcS9pjj> Record-Route: <sip:172.18.0.20:5099;r2=on;lr=on;ftag=cvcHy5UcS9pjj> Via: SIP/2.0/TLS X.X.X.11:443;branch=z9hG4bK24bf.52b1dd68e26010ba103bd8d6fdb53cf4.0 Via: SIP/2.0/UDP 172.18.0.40;received=172.18.0.40;rport=5060;branch=z9hG4bKH234Q58HpHNKK Max-Forwards: 66 From: "972522149596" <sip:FreeSWITCH@172.18.0.20:5099>;tag=cvcHy5UcS9pjj To: <sip:972587102881@172.18.0.20:5099> Call-ID: be55bd1e-537d-123a-65a9-0242ac120028 CSeq: 37932220 INVITE Contact: <sip:gw+kam-in@172.18.0.40:5060;transport=udp;gw=kam-in;alias=172.18.0.40~5060~1> User-Agent: FreeSWITCH-mod_sofia/1.10.6-release+git~20210325T131609Z~1ff9d0a60e~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 567 X-SESSION-ID: 721533765 X-INCOMING: PSTN X-from-freeswitch: +972522149596 X-FS-Support: update_display,send_info Remote-Party-ID: "972522149596" <sip:972522149596@172.18.0.20:5099>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1624937621 1624937622 IN IP4 X.X.X.7 s=FreeSWITCH c=IN IP4 X.X.X.7 t=0 0 m=audio 30018 RTP/AVP 102 101 a=rtpmap:102 opus/48000/2 a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=sendrecv a=rtcp:30019 a=ptime:20 a=ice-ufrag:bKPZt0dW a=ice-pwd:H4qyL7mPUOvoiAjosiLasXAdzj a=candidate:41sXGEsTafxHi8ms 1 UDP 2130706431 X.X.X.7 30018 typ host a=candidate:41sXGEsTafxHi8ms 2 UDP 2130706430 X.X.X.7 30019 typ host Response Device->Kamailio SIP/2.0 200 OK Via: SIP/2.0/TLS X.X.X.11:443;received=X.X.X.11;branch=z9hG4bK24bf.52b1dd68e26010ba103bd8d6fdb53cf4.0 Via: SIP/2.0/UDP 172.18.0.40;rport=5060;received=172.18.0.40;branch=z9hG4bKH234Q58HpHNKK Record-Route: <sip:X.X.X.11:443;transport=tls;lr;r2=on;ftag=cvcHy5UcS9pjj> Record-Route: <sip:172.18.0.20:5099;lr;r2=on;ftag=cvcHy5UcS9pjj> Call-ID: be55bd1e-537d-123a-65a9-0242ac120028 From: "972522149596" <sip:FreeSWITCH@172.18.0.20>;tag=cvcHy5UcS9pjj To: <sip:972587102881@172.18.0.20>;tag=5f75877e-58c2-44f3-a5b3-43f8786c42a1 CSeq: 37932220 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:972587102881@79.179.1.2:56308;transport=TLS;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 348 v=0 o=- 3833960833 3833960834 IN IP4 192.168.1.93 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 96 120 c=IN IP4 192.168.1.93 b=TIAS:96000 a=rtcp:4003 IN IP4 192.168.1.93 a=sendrecv a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:120 telephone-event/48000 a=fmtp:120 0-16 a=ssrc:816694098 cname:72cdb5f53fc2e8fb Kamailio NAT Detect/Manage Routes # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT xinfo("NATDETECT: Adding the rport parameter to the first Via header of the received message\n"); force_rport(); xinfo("NATDETECT: Trying to guess if client's request originated behind a nat. Testing 19.\n"); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xinfo("NATDETECT: REGISTER request. Creating a URI consisting of the source IP, port, and protocol and storing the URI in an Attribute-Value-Pair.\n"); fix_nated_register(); } else { if(is_first_hop()) { xinfo("NATDETECT: Adding an ;alias=ip~port~transport parameter to the contact URI containing the received ip, port, and transport protocol.\n"); set_contact_alias(); } } xinfo("NATDETECT: Setting flag FLT_NATS.\n"); setflag(FLT_NATS); } else { xinfo("NATDETECT: Didn't detect natted request by any of the given heuristics\n"); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xinfo("NATMANAGE: This is a request with to-tag that has 'nat=yes', setting branch flag FLB_NATB.\n"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xinfo("NATMANAGE: No NAT flag detected for this request\n"); return; } else { xinfo("NATMANAGE: NAT flag detected for this request\n"); } if(!is_present_hf("x-purpose")) { xinfo("NATMANAGE: x-purpose header does not appear in this request, SDP is searched for occurrence of RFC1918 or RFC6598 addresses.\n"); if(nat_uac_test("8")) { xinfo("NATMANAGE: Found occurrence of RFC1918 or RFC6598 addresses. Checking if there is a match of source address or uri with an address in the group of the dispatcher-list.\n"); if(ds_is_from_list()) { xinfo("NATMANAGE: There is a match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin direction=priv direction=pub"); } else { xinfo("NATMANAGE: No match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin direction=pub direction=priv"); } } else { xinfo("NATMANAGE: No occurrence of RFC1918 or RFC6598 addresses. Checking if there is a match of source address or uri with an address in the group of the dispatcher-list.\n"); if(ds_is_from_list()) { xinfo("NATMANAGE: There is a match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin trust-address direction=priv direction=pub"); } else { xinfo("NATMANAGE: No match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin trust-address direction=pub direction=priv"); } } } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xinfo("NATMANAGE: This is a request without to-tag and top-executed route block is branch_route, adding record route param ;nat=yes.\n"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xinfo("NATMANAGE: Adding an ;alias=ip~port~transport parameter to the contact URI containing the received ip, port, and transport protocol.\n"); set_contact_alias(); } } } #!endif return; } RTPEngine Logs 2021-06-29T13:07:05.440658779Z [1624972025.439547] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Received command 'offer' from 172.18.0.20:53880 2021-06-29T13:07:05.440690853Z [1624972025.439641] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Dump for 'offer' from 172.18.0.20:53880: { "supports": [ "load limit" ], "sdp": "v=0 2021-06-29T13:07:05.440695330Z o=FreeSWITCH 1624937621 1624937622 IN IP4 172.18.0.40 2021-06-29T13:07:05.440698353Z s=FreeSWITCH 2021-06-29T13:07:05.440700926Z c=IN IP4 172.18.0.40 2021-06-29T13:07:05.440703539Z t=0 0 2021-06-29T13:07:05.440706179Z m=audio 34404 RTP/AVP 102 101 2021-06-29T13:07:05.440708722Z a=rtpmap:102 opus/48000/2 2021-06-29T13:07:05.440711294Z a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 2021-06-29T13:07:05.440714053Z a=rtpmap:101 telephone-event/48000 2021-06-29T13:07:05.440716746Z a=fmtp:101 0-16 2021-06-29T13:07:05.440719258Z a=ptime:20 2021-06-29T13:07:05.440721823Z ", "ICE": "force", "direction": [ "priv", "pub" ], "replace": [ "session-connection", "origin" ], "call-id": "be55bd1e-537d-123a-65a9-0242ac120028", "received-from": [ "IP4", "172.18.0.40" ], "from-tag": "cvcHy5UcS9pjj", "command": "offer" } 2021-06-29T13:07:05.441069790Z [1624972025.439694] NOTICE: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] Creating new call 2021-06-29T13:07:05.441088698Z [1624972025.439809] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Default sink codec is opus/48000/2 2021-06-29T13:07:05.441091829Z [1624972025.439815] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Output DTMF payload type is 101 2021-06-29T13:07:05.441094966Z [1624972025.439821] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for opus/48000/2 2021-06-29T13:07:05.441097777Z [1624972025.439828] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec opus/48000/2 2021-06-29T13:07:05.441100540Z [1624972025.439833] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for opus/48000/2 2021-06-29T13:07:05.441103390Z [1624972025.439839] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for telephone-event/48000 2021-06-29T13:07:05.441106153Z [1624972025.439844] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec telephone-event/48000 2021-06-29T13:07:05.441108913Z [1624972025.439849] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for telephone-event/48000 2021-06-29T13:07:05.441111679Z [1624972025.440116] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 172.18.0.40:34404 2021-06-29T13:07:05.441124990Z [1624972025.440124] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 172.18.0.40:34405 2021-06-29T13:07:05.441663481Z [1624972025.441074] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Replying to 'offer' from 172.18.0.20:53880 (elapsed time 0.001414 sec) 2021-06-29T13:07:05.441694377Z [1624972025.441103] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Response dump for 'offer' to 172.18.0.20:53880: { "sdp": "v=0 2021-06-29T13:07:05.441701068Z o=FreeSWITCH 1624937621 1624937622 IN IP4 X.X.X.7 2021-06-29T13:07:05.441706421Z s=FreeSWITCH 2021-06-29T13:07:05.441711414Z c=IN IP4 X.X.X.7 2021-06-29T13:07:05.441716274Z t=0 0 2021-06-29T13:07:05.441721177Z m=audio 30018 RTP/AVP 102 101 2021-06-29T13:07:05.441726204Z a=rtpmap:102 opus/48000/2 2021-06-29T13:07:05.441731246Z a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 2021-06-29T13:07:05.441736873Z a=rtpmap:101 telephone-event/48000 2021-06-29T13:07:05.441742017Z a=fmtp:101 0-16 2021-06-29T13:07:05.441827012Z a=sendrecv 2021-06-29T13:07:05.441844744Z a=rtcp:30019 2021-06-29T13:07:05.441848881Z a=ptime:20 2021-06-29T13:07:05.441852444Z a=ice-ufrag:bKPZt0dW 2021-06-29T13:07:05.441856394Z a=ice-pwd:H4qyL7mPUOvoiAjosiLasXAdzj 2021-06-29T13:07:05.441860642Z a=candidate:41sXGEsTafxHi8ms 1 UDP 2130706431 X.X.X.7 30018 typ host 2021-06-29T13:07:05.441864955Z a=candidate:41sXGEsTafxHi8ms 2 UDP 2130706430 X.X.X.7 30019 typ host 2021-06-29T13:07:05.441868967Z ", "result": "ok" } 2021-06-29T13:07:07.001082810Z [1624972027.000176] DEBUG: [core] timer run time = 0.000029 sec 2021-06-29T13:07:07.817899628Z [1624972027.816949] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] >>> in_ssrc changed for: :0 new: 30adc352 2021-06-29T13:07:07.817932021Z [1624972027.816987] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] >>> out_ssrc changed for 172.18.0.40:34405 new: 30adc352 2021-06-29T13:07:07.817936388Z [1624972027.816998] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 201 2021-06-29T13:07:07.817939853Z [1624972027.817025] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] RR from 30adc352 about 0: FL 0 TL 0 HSR 0 J 0 LSR 0 DLSR 0 2021-06-29T13:07:07.817942689Z [1624972027.817030] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] No payload type known for RTCP RR, discarding 2021-06-29T13:07:07.817945696Z [1624972027.817036] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 202 2021-06-29T13:07:07.817948500Z [1624972027.817124] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] Forward to sink endpoint: 172.18.0.40:34405 2021-06-29T13:07:07.818762433Z [1624972027.817995] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 201 2021-06-29T13:07:07.818790347Z [1624972027.818031] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] RR from 30adc352 about 0: FL 0 TL 0 HSR 0 J 0 LSR 0 DLSR 0 2021-06-29T13:07:07.818794422Z [1624972027.818035] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] No payload type known for RTCP RR, discarding 2021-06-29T13:07:07.818797834Z [1624972027.818042] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 202 2021-06-29T13:07:07.818801095Z [1624972027.818122] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] Forward to sink endpoint: 172.18.0.40:34405 2021-06-29T13:07:07.847992398Z [1624972027.846284] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Received command 'answer' from 172.18.0.20:47817 2021-06-29T13:07:07.848048472Z [1624972027.846325] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Dump for 'answer' from 172.18.0.20:47817: { "supports": [ "load limit" ], "sdp": "v=0 2021-06-29T13:07:07.848058346Z o=- 3833960833 3833960834 IN IP4 192.168.1.93 2021-06-29T13:07:07.848063378Z s=pjmedia 2021-06-29T13:07:07.848067785Z b=AS:117 2021-06-29T13:07:07.848071736Z t=0 0 2021-06-29T13:07:07.848075782Z a=X-nat:0 2021-06-29T13:07:07.848079588Z m=audio 4002 RTP/AVP 96 120 2021-06-29T13:07:07.848084210Z c=IN IP4 192.168.1.93 2021-06-29T13:07:07.848089110Z b=TIAS:96000 2021-06-29T13:07:07.848092756Z a=rtcp:4003 IN IP4 192.168.1.93 2021-06-29T13:07:07.848096578Z a=sendrecv 2021-06-29T13:07:07.848100173Z a=rtpmap:96 opus/48000/2 2021-06-29T13:07:07.848104153Z a=fmtp:96 useinbandfec=1 2021-06-29T13:07:07.848108660Z a=rtpmap:120 telephone-event/48000 2021-06-29T13:07:07.848112776Z a=fmtp:120 0-16 2021-06-29T13:07:07.848116856Z a=ssrc:816694098 cname:72cdb5f53fc2e8fb 2021-06-29T13:07:07.848120776Z ", "ICE": "force", "direction": [ "pub", "priv" ], "replace": [ "session-connection", "origin" ], "call-id": "be55bd1e-537d-123a-65a9-0242ac120028", "received-from": [ "IP4", "79.179.1.2" ], "from-tag": "cvcHy5UcS9pjj", "to-tag": "5f75877e-58c2-44f3-a5b3-43f8786c42a1", "command": "answer" } 2021-06-29T13:07:07.848128079Z [1624972027.846411] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Default sink codec is opus/48000/2 2021-06-29T13:07:07.848132255Z [1624972027.846418] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Output DTMF payload type is 120 2021-06-29T13:07:07.848136164Z [1624972027.846424] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for opus/48000/2 2021-06-29T13:07:07.848139819Z [1624972027.846429] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec opus/48000/2 2021-06-29T13:07:07.848143930Z [1624972027.846434] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for opus/48000/2 2021-06-29T13:07:07.848169208Z [1624972027.846441] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for telephone-event/48000 2021-06-29T13:07:07.848174300Z [1624972027.846445] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec telephone-event/48000 2021-06-29T13:07:07.848178752Z [1624972027.846449] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for telephone-event/48000 2021-06-29T13:07:07.848183460Z [1624972027.846586] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 192.168.1.93:4002 2021-06-29T13:07:07.848188805Z [1624972027.846593] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 192.168.1.93:4003 2021-06-29T13:07:07.848193111Z [1624972027.846597] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [ice] Shutting down ICE agent (nothing to do) 2021-06-29T13:07:07.848197042Z [1624972027.847228] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Replying to 'answer' from 172.18.0.20:47817 (elapsed time 0.000890 sec) 2021-06-29T13:07:07.848201623Z [1624972027.847243] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Response dump for 'answer' to 172.18.0.20:47817: { "sdp": "v=0 2021-06-29T13:07:07.848205754Z o=- 3833960833 3833960834 IN IP4 172.18.0.50 2021-06-29T13:07:07.848209696Z s=pjmedia 2021-06-29T13:07:07.848213446Z b=AS:117 2021-06-29T13:07:07.848217361Z t=0 0 2021-06-29T13:07:07.848220816Z a=X-nat:0 2021-06-29T13:07:07.848224363Z m=audio 30018 RTP/AVP 96 120 2021-06-29T13:07:07.848228432Z c=IN IP4 172.18.0.50 2021-06-29T13:07:07.848232603Z b=TIAS:96000 2021-06-29T13:07:07.848235992Z a=ssrc:816694098 cname:72cdb5f53fc2e8fb 2021-06-29T13:07:07.848240381Z a=rtpmap:96 opus/48000/2 2021-06-29T13:07:07.848243923Z a=fmtp:96 useinbandfec=1 2021-06-29T13:07:07.848247725Z a=rtpmap:120 telephone-event/48000 2021-06-29T13:07:07.848386453Z a=fmtp:120 0-16 2021-06-29T13:07:07.848395759Z a=sendrecv 2021-06-29T13:07:07.848399822Z a=rtcp:30019 2021-06-29T13:07:07.848403748Z a=ptime:20 2021-06-29T13:07:07.848407203Z a=ice-ufrag:4Q9H38CE 2021-06-29T13:07:07.848410827Z a=ice-pwd:OR8EfUn5dCLbhZqyiwsTUOyUdW 2021-06-29T13:07:07.848414680Z a=candidate:f75EKiqnFiVfQdoZ 1 UDP 2130706431 172.18.0.50 30018 typ host 2021-06-29T13:07:07.848419247Z a=candidate:f75EKiqnFiVfQdoZ 2 UDP 2130706430 172.18.0.50 30019 typ host 2021-06-29T13:07:07.848432370Z ", "result": "ok" } 2021-06-29T13:07:07.872251466Z [1624972027.871342] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] Created candidate pair f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 between 172.18.0.50 and 172.18.0.40:34404, type prflx 2021-06-29T13:07:07.872276197Z [1624972027.871381] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] Triggering check for f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 2021-06-29T13:07:07.872286600Z [1624972027.871396] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] ICE pair f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 has been nominated by peer 2021-06-29T13:07:07.872289762Z [1624972027.871401] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] ICE not completed yet and no usable candidates 2021-06-29T13:07:07.872292668Z [1624972027.871407] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Successful STUN binding request from 172.18.0.40:34404 2021-06-29T13:07:07.919668940Z [1624972027.918862] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] >>> in_ssrc changed for: 172.18.0.40:34404 new: 70f328c1 2021-06-29T13:07:07.919698433Z [1624972027.918897] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] >>> out_ssrc changed for 192.168.1.93:4002 new: 70f328c1 2021-06-29T13:07:07.919703321Z [1624972027.918911] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64423 TS 2880) 2021-06-29T13:07:07.939554198Z [1624972027.938733] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64424 TS 3840) 2021-06-29T13:07:07.959769679Z [1624972027.959017] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64425 TS 4800) 2021-06-29T13:07:07.979861917Z [1624972027.979067] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64426 TS 5760) 2021-06-29T13:07:07.999566603Z [1624972027.998777] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64427 TS 6720) 2021-06-29T13:07:08.019668998Z [1624972028.018802] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64428 TS 7680) 2021-06-29T13:07:08.040856248Z [1624972028.039663] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64429 TS 8640) 2021-06-29T13:07:08.061133612Z [1624972028.059847] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64430 TS 9600) Edward

1 1

Guidance building a WebRTC to Asterisk Intermediate Proxy
by Peter Manley 25 Jun '21

25 Jun '21

Hello, I'm working on building an RTP Proxy which will allow both regular SIP to SIP transactions and WebRTC to SIP transactions to an Asterisk Server. SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Asterisk Server AND WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Asterisk Server The Kamailio/RTPEngine Proxy is stateless (does not use the tm or registerer modules), and uses the Path module to replace the Route header with a Path header. It works grep for SIP/UDP to SIP/UDP transactions. HOWEVER... It only partially works for WSS to SIP/UDP transactions. The REGISTER/401(Unauthorized), REGISTER/200(OK) transaction works well. When the Asterisk server sends a SIP Options message, I get the following error: Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: <script>: => OPTIONS < sip:l4umfumh@192.0.2.97;transport=ws;alias=209.141.117.212~59751~6 -> <null> (IP:172.16.0.48:sp) Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> [core/msg_translator.c:2815]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> [core/msg_translator.c:2003]: build_req_buf_from_sip_req(): could not create Via header Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> [core/forward.c:550]: forward_request(): building failed Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> [core/receive.c:425]: receive_msg(): error while trying script It appears that the WebSocket corresponding to the WSS leg cannot be located when the request is initiated from the Asterisk Server. Any thoughts on how I can get this to work in the other direction (Asterisk to Kamailio)? I've been looking at the WebRTC to SIP Client and Server (https://github.com/havfo/WEBRTC-to-SIP), but that doesn't quite do what I want it to do (i.e. it appears to act as a registrar server and doesn't forward REGISTER requests on to another registrar). Thanks and regards, Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

2 1

using both ldap and local authenticate
by Thành Trần Duy 25 Jun '21

25 Jun '21

Hi all, I am using kamailio as PBX with some softphones and hard phones. I want user on a pc to login with LDAP and users using a hard phone login normally (username/password). Can I use 2 types of authentication at the same time? Many thanks, Tran Duy Thanh

2 2

Possible memory leak on 5.5.x (new)?
by George Diamantopoulos 25 Jun '21

25 Jun '21

Hello all, I'm still investigating the (most likely non-kamailio-related) memory leak of my previous message to the list, there have been no developments so far. I'll update if anything changes. This concerns a new finding which seems to affect kamailio 5.5.x. I have two kamailio instances receiving the same traffic via round-robin. I upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared of any issues arising. I was lucky to do so, because with identical configuration, 5.5.x seems to run out of SHM very quickly. Here are links to graphs produced by our monitoring system: Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png The configuration uses mtree, htable, vars and vns extensively. Has anyone come across anything similar? Let me know if I can provide any further information to help disect this. Thanks! BR, George

2 1

rr module: record_route_advertised_address()/record_route_preset() do ignore custom user AVP
by Valentin Christoph 25 Jun '21

25 Jun '21

Hi, Maybe a discussion for the DEV mailing list???? Want to overwrite the FQDN by a NATed IP address in the topmost Record-Route of an outgoing INVITE request. Cannot set the advertise parameter (this must tell the FQDN for all other situations), only in some special situations I want to overwrite the FQDN in Record-Route. My solution would be to implement in the Kamailio source code (rr module) that not only record_route() uses the get_custom_user() function, but the other both functions uses it, too. Any better proposals? All the best Christoph

2 3

Kamailio $Rp variable showing wrong port
by Shahid Hussain 25 Jun '21

25 Jun '21

Hi, I have configured the dispatcher default socket as below IP , the expectation here is that kamailio should send all traffic from this IP (port will be ephemeral) over TCP. modparam("dispatcher", "ds_default_socket", "tcp:172.27.45.219:5060") This system also listens on port 8080 which is meant for websocket communication (dispatcher communication is over non websocket). Kamailio is sending an OPTION message to dispatcher IP over TCP on 5060 and getting 200 OK response but Kamailio considering the response is on websocket port 8080 ,$Rp value is 8080. onreply_route { if (($Rp == 8080 ) && !(proto == WS)) { xlog("L_WARN", "SIP response received on $Rp protocol:proto RIP:$Ri SIP:$si method:$rm cid:$ci replystatus:$rs len:$ml\n"); xlog("L_WARN", "SIP message received: $mb\n"); drop; } ..... .... } Following is the output of above logs: Jun 24 15:10:08 localhost /usr/sbin/kamailio[6223]: WARNING: {2 10 OPTIONS 1ab6573a4640823b-6209(a)172.27.45.198} <script>: SIP response received on 8080 protocol:tcp RIP:172.27.45.219 SIP:172.27.44.125 method:OPTIONS cid:1ab6573a4640823b-6209@172.27.45.198 replystatus:200 len:416 Jun 24 15:10:08 localhost /usr/sbin/kamailio[6223]: WARNING: {2 10 OPTIONS 1ab6573a4640823b-6209(a)172.27.45.198} <script>: SIP message received: SIP/2.0 200 OK#015#012Via: SIP/2.0/TCP 172.27.45.198:5060;branch=z9hG4bK4526.92b79f34000000000000000000000000.0;received=172.27.45.219#015#012From: <sip:dispatcher@localhost>;tag=1ae6c95d92f9ea9ce927f715de5c48c8-eb141029#015#012To: <sip:172.27.44.125:5060;transport=tcp>;tag=a9d542e0#015#012Call-ID: 1ab6573a4640823b-6209(a)172.27.45.198#015#012CSeq: 10 OPTIONS#015#012Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, NOTIFY#015#012Content-Length: 0 Also I have attached a pcap. Dispatcher.list configuration: 1 sip:172.27.44.125:5060;transport=tcp 0 0 duid=sample-cas;maxload=1000 Thanks, Shahid

3 10

S-CSCF and I-CSCF
by Jayanth Acharya 24 Jun '21

24 Jun '21

Hi, Have found example configuration for S-CSCF, I-CSCF and P-CSCF in Kamailio repository, and a S-CSCF module for the registered user data, but no recent updates. Wondering if anyone can share insights into how widely it has been used, or interops done in say telco networks with it ? Does it handle third-party registrations, is it actually able to do a Cx interaction with HSS ? Finally, how have these components that make up the IMS-core from SIP signaling standpoint, been tested ? Are there any signaling only test harnesses ? cheers, Jay

2 1

Errors after upgrade to 5.5
by Daniel W. Graham 23 Jun '21

23 Jun '21

Receiving constant errors for all functions after upgrade to 5.5, any change that could have caused this? ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:968] took too long [1251 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:988] took too long [1309 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:1038] took too long [1321 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (5)] cfg [/usr/local/etc/kamailio/kamailio.cfg:452] took too long [1342 us] ERROR: <core> [db_query.c:68]: db_do_submit_query(): alert - query execution too long [1493 us] for [select `password`,`id` from `subscriber` where `username`='x'] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [www_authenticate (26)] cfg [/usr/local/etc/kamailio/kamailio.cfg:951] took too long [1667 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:968] took too long [1694 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:988] took too long [1817 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:1038] took too long [1835 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (5)] cfg [/usr/local/etc/kamailio/kamailio.cfg:452] took too long [1869 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [www_authenticate (26)] cfg [/usr/local/etc/kamailio/kamailio.cfg:951] took too long [1405 us] ……. -dan

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sr-users June 2021