sr-users September 2020

sr-users@lists.kamailio.org

84 participants
106 discussions

by Pavithra Mohanraja

Hi, Kindly give me some guidelines on how to convert diameter messages to http messages when establishing connection between PCSCF and PCF Kindly help Thanks, Pavithra

3 years, 7 months

Few newbie questions

by Adam Rutkowski

Hi Everyone As I am just dipping my foot in the VOIP waters so please bear with me. Although it may be more appropriate for Freeswitch group I am sure there are plenty of knowledgeable folks here. I'm playing with freeswitch in my lab environment and really like what Kamailio has got to offer. That being said, there are few concepts I need help understanding. Consider this ( for hosted services I suppose ) [image: Screenshot 2020-09-16 at 22.21.22.png] 1 Does this topology seem right ? 2 If Kamailio is used as registar and load balancer, does that mean all domains, extensions must be present on all the FS boxes ? 3 How will auto provisioning work in this case ? Now, in the below scenario, clients have got FS box deployed on prem [image: Screenshot 2020-09-16 at 22.28.09.png] 4 Is there any reason to keep the FS farm ? or Kamailio poxying calls to VOIP providers is enough ? 5 What roles Kamailio could be used for in this case ? 6 Is deploying on prem PBX for a customer a popular solution ? 7 Anyone deployed above on Docker/ Kubernetes ? How's the performance and the bad bits if any ? I do realize this is pretty basic stuff for everyone in here, however, I would appreciate any helpful comments and if there is some good literature/ links you can recommend, I am all ears. Regards Adam

3 years, 7 months

Kamailio IMS Diameter Messages

by Pavithra Mohanraja

Hi, Is there any possibility of converting diameter messages to http requests in IMS. Since I am working in 5g based network, In order to connect pcscf to pcf, Rx interface has to be established. But PCF is not accepting diameter messages. It is failing with error "No connected DefaultRoute peer found for app_id 16777236 and vendor id 10415" I used IMS Diameter server since it is mentioned as converting diameter messages to http json messages. But even before calling that diameter server, It is failing with App_id issue. How it works? Need to understand, how the diameter messages are working since in 5g , everything is based on http rest apis. Is there any possibility of connecting to it or there is no support in kamailio IMS for 5g. If you have any idea, please let me know. It will be very useful for me. Thanks, Pavithra

3 years, 7 months

multi netwotk interface kamailio stops processing packets in one of it's network interfaces although keeps processing in the others

by Nuno Miguel Reis

Hello everyone. I've started to have an issue with a kamailio 4.4 instance listening on multiple network interfaces where it stops processing SIP on one of it's network interfaces but still continues to work fine on the others. If I restart kamailio everything starts working fine again. I'm using the default 'children=8' and one of my guesses on why this could be happening is that the number of childs processing couldn't be enough. Do you remember anything else on an issue like this where I should be looking? Thanks. -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

3 years, 7 months

Regular Expression Evaluation

by Mack Hendricks

I already run into issues with this. If I want to only execute a statement if it’s an INVITE with a request domain of 10.10.*.* how do I format? Below is what I have. I always have to play with it a bit. Looking to get a better understanding. The regular expression tester I’m using states that it’s valid. What am I doing wrong? if ((is_method("INVITE")) && ($rd =~ "10\.10\..+")) { # Do something xlog("fix got a 10.10 address"); } Thanks

3 years, 7 months

CockroachDB and Kamailio

by Noah Mehl

Has anyone been down this path before? We are trying to test this out and the results are pretty promising so far. I realize the lack of Stored Procedures and Triggers make this untenable for many Postgres based implementations. Thanks! ~Noah

3 years, 7 months

Kamailio does not forward BYE to Asterisk

by egemen ulus

Hi all, I have an infrastructure with Kamailio (reg), Asterisks farm, and Kamailio (4.3) proxy which dispatches calls. Trunk-1 (Vlan_A) Kam(reg) <> Asterisk Farm (5) <> Kam(proxy) <> Trunk-2 (Vlan_B) Trunk-3 (Vlan_C) For each trunk, I have to use the IP of the VLAN interface in "contact header". That means when I need to send a call to Trunk-1 I have to set Vlan_A IP address on the contact header. I have achieved that by rewriting Contact Header. ### remove_hf("Contact:"); append_hf("Contact: <sip:$fU@$var(recieved)>\r\n", "Call-ID"); #by using db ### But this causes a problem, Kamailio does not forward the BYE message that comes from Trunk-X to Asterisk (In before, the BYE message had Asterisk IP in R-URI). However, when I use topoh module it forward properly for one trunk. But I can not set multiple mask IP in topoh module. I'm using the dialog module and I thought it can handle all requests in one dialog to forward to the correct asterisk but it did not. Is there any suggestion for this case? And also how many sub-interfaces can Kamailio handle for a good performance? Or do you suggest multiple Kamailio instances instead of multiple sub-interfaces on one server? [https://ipmcdn.avast.com/images/icons/icon-envelope-tick-round-orange-anima…]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>

3 years, 7 months

Kamailio used as proxy for Conference factories

by Adrian Tabacioiu

Hello, I am new in this VOIP field, yet I have to make a decision in using a SIP proxy that would allow the configuration of multiple conference factories, make load balancing when creating new conference, and provide presence information related to active conferences. Can you please tell me if this is possible with Kamailio, and give me some reference to a module or some documentation that would relate to that ? Thank you in advance. Best regards, Adrian Tabacioiu

3 years, 7 months

Proxying BFCP

by René Hansen

Hi, I'm rather new to Kamailio and RTPEngine, so I'm looking for some help in locating any insights, documentation, guides or examples that resemble my use case. Currently I've got both running in working order and I can do SIP calls with both video and audio. Now I'm trying to figure out how to pass through BFCP as well. From what I understand so far, since BFCP isn't strictly RTP, it's not something that's outright supported. Please correct me if my assumption is wrong. My observation is that "*m=application PORT {UDP,TCP}/BFCP **" SDP lines doesn't seem to open for traffic as other media lines do. (Obviously expected if it's not supported) So I'm really just looking for any suggestions for, or where to start looking, in getting BFCP support. I do have the option of BFCP over both TCP and UDP, so if I could somehow include the originator IP as an added attribute in the re-written SDP answer, I could then just connect directly, around the proxy. I've yet to figure out how to do that as well however. Any pointers are much appreciated. Regards, René Hansen

3 years, 7 months

Called with SDP without DTLS fingerprint from WebRTC to SIP

by Benjamin Flügel | vio:networks

Hey guys and girls, I'm trying to configure Kamailio so that I can communicate over it using WebRTC and SIP. Client 1 is a softphone based on SIPJS. Client 2 is a Yealink phone. Both are registered with the Kamailio. Behind the Kamailio there is another Asterisk. The Kamailio acts as a proxy here. So far everything works. Only if I accept the call, which is made from Client 1 to Client 2 or vice versa, then I get this error in Client 1: Called with SDP without DTLS fingerprint. I can see that the fingerprint is present in the INVITE from Client 1 to Client 2. However, it is missing in the OK. The lower part contains my NATMANAGE config. This could probably be optimized a lot, but for now it should work. Unfortunately I have only recently started with Kamailio. I hope you can help me there. I can also send you the SIP-Log if you want to. Many thanks in advance. Greetings Benny route[NATMANAGE] { if (is_request()) { if(has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } } } if (!isbflagset(FLB_BRIDGE)) { return; } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS))) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { xlog("IS TRUSTED"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } if(isbflagset(FLB_RTPWS)){ xlog("IS WS"); if(is_request()){ xlog("IS REQUEST"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; rtpengine_manage($xavp(r=>$T_branch_idx)); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } } else { xlog("IS RESPONSE"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal loop-protect"; } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } rtpengine_manage($xavp(r=>$T_branch_idx)); } } else { xlog("SIP --> SIP"); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); rtpengine_manage(" direction=internal direction=external ICE=remove loop-protect"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } xlog("NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; }

3 years, 7 months

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sr-users September 2020