sr-users July 2023

sr-users@lists.kamailio.org

28 participants
32 discussions

by Duarte Rocha

Hello all, I'm currently doing some experiments with RTPENGINE and I need some help configuring it. First scenario i'm under NAT and Kamailio receives an INVITE with a private address, rewrites (c) line on SDP with it's own IP address and relays the message to the destination endpoint. After that the originator receives the 183 Session Progress with kamailio's IP as (c) parameter. After receiving the first RTP packet from the originator the media streams flow normally and I have sound on both sides including ringing. Second scenario I use different equipment that is not under NAT, but the remaining scenario is the same, except the originator doesn't send an RTP packet after the 183 Session Progress. However, in this case, no RTP stream is relayed to the originator and as a consequence it doesn't hear a ring or voice when the call is answered. As i see it, it looks like Kamailio is on passive mode to the A-Leg of the call and waits for a RTP packet to establish its stream. Is this correct? If so, how do i change this behaviour? Cheers

9 months, 2 weeks

Re: Kamailio as SIP proxy for an Asterisk with Private IP Address

by Lewis Hutchinson

This is always a good place to start https://www.fredposner.com/1457/kamailio-behind-nat/ Lewis Mission Labs Limited is registered in England, company number 10040088. Trading Office: The Old Milk Depot, Bacup Rd, Rossendale, BB4 7FE. Registered office: The Scalpel, 18th Floor 52 Lime Street, London, EC3M 7AF. Email confidentiality notice: This message is private and confidential. If you have received this message in error, please notify us and remove it from your system. Please consider the environment before you print this email.

9 months, 2 weeks

tcpops: Variable conid on tcp-closed event

by Michal Hajek

Hi, I have problem with the conid variable, when I use in an event when closing a tcp connection. Simple kamailio.cfg: debug=3 log_stderror=no loadpath "/usr/lib64/kamailio/modules" memdbg=5 memlog=5 log_facility=LOG_LOCAL3 listen=udp:10.10.8.94:5060 listen=tcp:10.10.8.94:5060 loadmodule "sl.so" loadmodule "xlog.so" loadmodule "pv.so" loadmodule "tcpops.so" request_route { xlog("L_INFO", "Welcome TCP conn $conid\n"); exit; } event_route[tcp:closed] { xlog("L_INFO", "Close TCP conn $conid\n"); } Log: kamailio DEBUG: <core> [core/ip_addr.c:577]: print_ip(): tcpconn_new: new tcp connection: 10.10.9.64 kamailio DEBUG: <core> [core/tcp_main.c:1216]: tcpconn_new(): on port 38124, type 2, socket 34 kamailio DEBUG: <core> [core/tcp_main.c:1549]: tcpconn_add(): hashes: 3487:2267:2872, 19 kamailio DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): processing io_watch_add(0xb39c40, 34, 2, 0x14c2479867f0) - fd_no=28 kamailio DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xb39c40, 34, -1, 0x0) fd_no=29 called kamailio DEBUG: <core> [core/tcp_main.c:4557]: handle_tcpconn_ev(): sending to child, events 2001 kamailio DEBUG: <core> [core/tcp_main.c:4227]: send2child(): selected tcp worker idx:2 proc:14 pid:1203550 for activity on [tcp:10.10.8.94:5060], 0x14c2479867f0 kamailio DEBUG: <core> [core/tcp_read.c:1735]: handle_io(): received n=8 con=0x14c2479867f0, fd=6 kamailio DEBUG: <core> [core/tcp_read.c:284]: tcp_read_data(): EOF on connection 0x14c2479867f0 (state: 3, flags: 4118) - FD 6, bytes 218, rd-flags 10000 ([10.10.9.64]:38124 -> [10.10.9.64]:5060) kamailio DEBUG: <core> [core/tcp_read.c:1533]: tcp_read_req(): content-length=0 kamailio DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 kamailio DEBUG: <core> [core/parser/msg_parser.c:678]: parse_msg(): SIP Request: kamailio DEBUG: <core> [core/parser/msg_parser.c:680]: parse_msg(): method: <OPTIONS> kamailio DEBUG: <core> [core/parser/msg_parser.c:682]: parse_msg(): uri: <sip:100@localhost> kamailio DEBUG: <core> [core/parser/msg_parser.c:684]: parse_msg(): version: <SIP/2.0> kamailio DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 kamailio DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKhjhs8ass877>; state=15 kamailio DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 kamailio DEBUG: <core> [core/parser/msg_parser.c:554]: parse_headers(): Via found, flags=2 kamailio DEBUG: <core> [core/parser/msg_parser.c:556]: parse_headers(): this is the first via kamailio DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Call-ID] type 6 kamailio DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [From] type 4 kamailio DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [To] type 3 kamailio DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=10 kamailio DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): <To> [20]; uri=[sip:100@localhost] kamailio DEBUG: <core> [core/parser/msg_parser.c:175]: get_hdr_field(): to body (20)[<sip:100@localhost> ], to tag (0)[] kamailio DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Content-Length] type 12 kamailio DEBUG: <core> [core/parser/msg_parser.c:186]: get_hdr_field(): content_length=0 kamailio DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [CSeq] type 5 kamailio DEBUG: <core> [core/parser/msg_parser.c:153]: get_hdr_field(): cseq <CSeq>: <63104> <OPTIONS> kamailio DEBUG: <core> [core/receive.c:389]: receive_msg(): --- received sip message - request - call-id: [ahoj423432432] - cseq: [63104 OPTIONS] kamailio DEBUG: {1 ahoj423432432} <core> [core/receive.c:261]: ksr_evrt_pre_routing(): event route core:pre-routing not defined kamailio DEBUG: {1 ahoj423432432} <core> [core/receive.c:471]: receive_msg(): preparing to run routing scripts... kamailio DEBUG: {1 ahoj423432432} <core> [core/tcp_main.c:1645]: _tcpconn_find(): found connection by id: 19 kamailio INFO: {1 ahoj423432432} <script>: Welcome TCP conn 19 kamailio DEBUG: {1 ahoj423432432} <core> [core/receive.c:529]: receive_msg(): request-route executed in: 47 usec kamailio DEBUG: {1 ahoj423432432} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/xavp.c:640]: xavp_destroy_list(): destroying xavp list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/xavp.c:640]: xavp_destroy_list(): destroying xavp list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/xavp.c:640]: xavp_destroy_list(): destroying xavp list (nil) kamailio DEBUG: {1 ahoj423432432} <core> [core/receive.c:628]: receive_msg(): cleaning up kamailio DEBUG: <core> [core/tcp_read.c:1634]: tcp_read_req(): EOF after reading complete request ([10.10.9.64]:38124 -> [10.10.9.64]:5060) kamailio DEBUG: <core> [core/tcp_read.c:1653]: release_tcpconn(): releasing con 0x14c2479867f0, state -1, fd=6, id=19 ([10.10.9.64]:38124 -> [10.10.9.64]:5060) kamailio DEBUG: <core> [core/tcp_read.c:1657]: release_tcpconn(): extra_data (nil) kamailio DEBUG: <core> [core/tcp_main.c:3653]: handle_tcp_child(): reader response= 14c2479867f0, -1 from 2 kamailio DEBUG: <core> [core/tcp_main.c:3578]: tcp_emit_closed_event(): TCP closed event creation triggered (reason: 0) kamailio DEBUG: tcpops [tcpops.c:295]: tcpops_handle_tcp_closed(): received TCP closed event kamailio DEBUG: tcpops [tcpops.c:249]: tcpops_tcp_closed_run_route(): event reason id: 0 rt: 1 kamailio INFO: <script>: Close TCP conn <null> Can you help me, where is issue?

9 months, 2 weeks

Kamailio as SIP proxy for an Asterisk with Private IP Address

by John shuku

Dear Kamailio users, Im new to kamailio so excuse me if the question doesnt make sense. I have an asterisk server with private IP address behind the NAT . Is it possible to have a Kamailio Server with Public IP Address as a proxy server for my asterisk ? What I want to achieve is that remote users should be able to call internal users who are registered in the Asterisk . The problem here is that Asterisk does not have public IP address and I want the remote users to be registered in my asterisk . Best WishesJ. sh

9 months, 2 weeks

Re: Add a digest authorization with password and username (INVITE)

by Max Kemel

Thank you very much Barry, that was it 🙂 Max Kemel

9 months, 3 weeks

Add a digest authorization with password and username (INVITE)

by Max Kemel

Hello, I would appreciate any help you can provide on this matter, I would like to make an outgoing INVITE request which will have a digest authorization with password and username. I was wandering around the Auth module to find the most fit method to build this header but hadn't found anything. Maybe I am missing something really obvious here, thanks for the help in advance Best regards, Max.

9 months, 3 weeks

Kamailio v5.5.7 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v5.5.7 stable release is out. This is a maintenance release of the stable branch 5.5 that includes fixes since the release of v5.5.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.7. Note that 5.5 is now the third last stable branch, v5.5.7 being the last planned released in 5.5.x series. The latest two stable branch are 5.6 and 5.7, with v5.7.1 being released a while ago. For more details about version 5.5.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/07/kamailio-v5-5-7-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 World Conference - www.kamailioworld.com

9 months, 3 weeks

Releasing Kamailio v5.5.7

by Daniel-Constantin Mierla

Hello, I am planning to release v5.5.7 our of branch 5.5 this week, most likely on Wednesday, July 19, 2023. It will mark the end of official packaging for release series 5.5.x, currently the latest two stable branches are 5.6 and 5.7. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - www.kamailioworld.com

9 months, 3 weeks

Re: WebRTC proxy with rtpengine - How to change from RTP to SRTP on 488 response

by Alex Balashov

I wouldn't do a delete in that situation; it might be simpler to just do another rtpengine_offer() on a <Call-ID, From-tag> tuple that RTPEngine already knows about, just as one would in a reinvite. I believe rtpengine_manage() would cover this situation as well. And yes, then just t_relay() anew! > On Jul 18, 2023, at 11:41 AM, Barry Flanagan <barry(a)flantel.com> wrote: > > On 18/07/2023 16:33, Alex Balashov wrote: >> Barry, >> >> Might it not be simpler to follow Richard's first suggestion in that thread, and simply catch the 488 response in a failure_route and re-issue a non-SRTP offer on another branch? > > hi Alex, > > Might be. I am just unsure about how to instruct rtpengine to re-send the plain RTP invite. Do I do a rtpengine_delete and then restart with rtpengine_manage() then relay again? > > Thanks. > > -Barry > > >> >> -- Alex >> >>> On Jul 18, 2023, at 11:05 AM, Barry Flanagan <barry(a)flantel.com> wrote: >>> >>> Hi, >>> >>> I have implemented a WebRTC proxy using Kamailio and rtpengine, in front >>> of Asterisk servers. I have an issue where I do not know in advance >>> whether the Asterisk client is set to use RTP or SRTP. >>> >>> My implementation expects the web side to be SRTP and the Asterisk side >>> plain RTP. However, if the asterisk side requires SRTP I receive a 488 >>> Not Acceptable here. The same is true when an SRTP client from the >>> Asterisk side dials out to the webrtc client, I need to catch whether >>> the request is SRTP or RTP and generate rtpengine offer accordingly. >>> >>> This was discussed in github, with a description of a solution here: >>> https://github.com/sipwise/rtpengine/issues/435#issuecomment-391937398 - >>> however I am unsure how to implement this. >>> >>> Does anyone have an example of how I can change the rtpengine SDP in >>> such a case? >>> >>> Thanks >>> -- >>> -Barry >>> __________________________________________________________ >>> Kamailio - Users Mailing List - Non Commercial Discussions >>> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >>> Important: keep the mailing list in the recipients, do not reply only to the sender! >>> Edit mailing list options or unsubscribe: > > -- > -Barry > -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

9 months, 3 weeks

WebRTC proxy with rtpengine - How to change from RTP to SRTP on 488 response

by Barry Flanagan

Hi, I have implemented a WebRTC proxy using Kamailio and rtpengine, in front of Asterisk servers. I have an issue where I do not know in advance whether the Asterisk client is set to use RTP or SRTP. My implementation expects the web side to be SRTP and the Asterisk side plain RTP. However, if the asterisk side requires SRTP I receive a 488 Not Acceptable here. The same is true when an SRTP client from the Asterisk side dials out to the webrtc client, I need to catch whether the request is SRTP or RTP and generate rtpengine offer accordingly. This was discussed in github, with a description of a solution here: https://github.com/sipwise/rtpengine/issues/435#issuecomment-391937398 <https://github.com/sipwise/rtpengine/issues/435#issuecomment-391937398>- however I am unsure how to implement this. Does anyone have an example of how I can change the rtpengine SDP in such a case? Thanks -- -Barry

9 months, 3 weeks

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sr-users July 2023