- sr-users - kamailio.org

VoLTE/IMS third party registration issues
by ruben.iniguez＠vodafone.com 23 May '24

23 May '24

Checking Kamailio in a VoLTE environment we are finding issues to make it work. First issues arise when getting AS to extract the required info from third party registration, primarily IMPUs. Third party registration is sent to AS with IMPI in To header, preventing the AS from extracting the IMPU from the most obvious place, the To header, according to 3GPP 24.229: "Notification of Application Servers about registration status [...] the To header field, which shall contain a non-barred public user identity belonging to the service profile of the processed Filter Criteria. It may be either a public user identity as contained in the REGISTER request received from the UE or one of the implicitly registered public user identities in the service profile, as configured by the operator;" On the other hand, the third party registration can only embed in its body either the original request or the original response but not both (notice mutually exclusive "else if" statements in third_party_reg.c). Both are needed to extract Feature-Caps (from original request) and IMPUs (for instance from P-Associated-URI in original response). Therefore to extract that info the AS should subscribe to the reg info in S-CSCF, but such SUBSCRIBE is rejected with 403. From Registrar IMS SCSCF documenation it looks like this is not supported: "This function checks to see that a SUBSCRIBE request is authorised to subscribe to the particular identity. Only 3 entities can subscribe: The user agent to its own state The P-CSCF specified in the path header for that user Application Server (AS) not yet implemented" Therefore question to the community is how these limitations are overcome in existing VoLTE deployments with Kamailio? Any advice?

1 0

Kamailio IPSec module
by H Yavari 23 May '24

23 May '24

Hi all, I was reviewing the `ims_ipsec_pcscf` code and noticed that this module creates a pool of sockets using different ports (ipsec_max_connections). I'm unclear on the necessity of this approach. Can't we simply create one server listener and one client listener to handle all UE connections? If this is feasible, is there still a need to create the pool at startup? I'm curious if this is due to an architectural limitation or if the IPSec module can be modified to replace the current implementation with a more efficient one. Thank you for your insights. Best regards,Hossein

3 5

S-CSCF SiFC support
by ruben.iniguez＠vodafone.com 22 May '24

22 May '24

Hello In the context of an IMS deployment, does anybody know if Shared iFC mechanism is supported in Kamailio? I don't see configuration options for it, and just wanted to confirm if only regular iFCs (residing on HSS) are supported. Thank you

1 0

Assistance Required for VoNR Setup with Version 5.8.1
by venkateshcoff＠gmail.com 22 May '24

22 May '24

Hi, I am trying to set up VoNR with version 5.8.1, but I cannot find the configuration information and documentation anywhere. Has anyone tested a VoNR call? Thanks.

1 0

usrloc: keepalives not being sent to UDP endpoints for a very long time
by luke＠primevox.net 21 May '24

21 May '24

We've noticed an issue with UDP endpoints and the usrloc module not sending keepalive packets to them for a long time. Eventually, usrloc will begin doing it, but sometimes it's anywhere between 10 minutes and multiple hours before the endpoint starts receiving an OPTIONS packet. TCP and TLS endpoints do not have this issue - they being receiving OPTIONS after registration + ka_timeout elapsed. Has anyone else seen this? It can be very easily seen with a pcap on a device registering via UDP.

1 0

A question about Db_flatstore Module in Kamailio
by فرزانه سلطان زاده 20 May '24

20 May '24

Hi, Hope to find you well, I have a question about the 'db_flatstore' module in Kamailio. Here's my script for log rotation: for f in acc_cdrs_*.log; do mv -- "$f" "${f%.log}.temp" done docker exec -it pcscf kamcmd flatstore.k_rotate rm *.temp I want to know if the kamcmd command is blocking and the files can be removed after that or if I should wait for it to be done before removing the renamed files. Regards, Farzaneh

3 2

Kamailio multidomain IMS
by Valentin Christoph 20 May '24

20 May '24

Dear IMS enthusiasts :-) Has anyone experience with Kamailio IMS (CSCF) setups, where users with different domain names can be registered at one and the same S-CSCF? Is this feasible at all? I mean one IMS for users <user>@ims1.example.com, <user>@ims2.example.com, <user>@ims3.example.com,........ in one and the same S-CSCF, I-CSCF, P-CSCF. Thanks, Christoph

2 1

q_malloc.c:520 WARNINGs
by Joel Serrano 20 May '24

20 May '24

Hi, Does anyone know what this WARNING means or how to fix it? 2024-03-09T00:39:15.406674-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406823-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406973-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407089-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407178-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407278-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407375-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) It happens after running dp_translate(). These are my settings: debug=L_DBG debug=L_NOTICE memdbg=L_INFO memlog=L_INFO corelog=L_ERR mem_summary=12 log_stderror=no log_facility=LOG_LOCAL0 log_name="csbc" log_prefix="{$mt $hdr(CSeq) $ci} " mem_join=1 Any ideas? Thanks, Joel.

2 5

tm.t_uac_wait ACK does not have Max-Forwards header
by Juha Heinanen 20 May '24

20 May '24

Kamailio sends INVITE by tm.t_uac_wait RPC command: INVITE sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com> From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Content-Length: 131 User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Max-Forwards: 10 Contact: <sip:click2dial@192.168.187.215:5060> Content-Type: application/sdp .. sdp omitted UAS accepts the INVITE: SIP/2.0 200 Answering Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Server: baresip v3.12.0 (x86_64/Linux) Contact: <sip:test-0x55e7f1f78e70@192.168.187.215:5022> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,UPDATE,REFER Supported: gruu,replaces,norefersub Content-Type: application/sdp Content-Length: 239 .. sdp omitted After receiving the response, Kamailio automatically sends ACK: ACK sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 Call-ID: 6649e78355ac2(a)test.tutpro.com To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e CSeq: 1 ACK User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Content-Length: 0 The problem is that Max-Forwards header is missing from the ACK. Should I have something in config file to add it or is this a bug? -- Juha

1 3

Segfault on startup when using TLS in kamailio 5.7.4
by stefanr＠wave.com 17 May '24

17 May '24

Hi there, We are encountering consistent segfaults after rebooting our Kamailio instance with incoming traffic, specifically when using Kamailio 5.7.4. We think this issue did not occur with version 5.7.2, so it seems to have been introduced in either 5.7.3 or 5.7.4. Due to team bandwidth constraints and the potential impact on production traffic, we don't want to spend time on trying to reproduce the issue. So we have decided to downgrade to 5.6.4, which we confirmed to be stable. (Probably 5.7.2 would be too - but we didn't try). Unfortunately, our logging was only set to WARNING level, and we did not capture a core dump, so we cannot provide additional details beyond the following logs: This was with tcp_reuse_ports=yes: 2024-05-17T15:42:55.582475541Z Listening on 2024-05-17T15:42:55.582512370Z [redacted] 2024-05-17T15:42:55.582538161Z tls: 10.X.X.X:5061 advertise Y.Y.Y:5061 2024-05-17T15:42:55.582543750Z Aliases: 2024-05-17T15:42:55.582549081Z tls: [redacted]:5061 2024-05-17T15:42:55.582574890Z 2024-05-17T15:42:55.587876630Z 0(1) WARNING: tls [tls_init.c:978]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 18874368 and 9437184 bytes 2024-05-17T15:42:55.703927049Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 23 2024-05-17T15:42:55.703972029Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 15 exited by a signal 11 2024-05-17T15:42:55.703978409Z 0(1) ALERT: <core> [main.c:795]: handle_sigs(): core was generated 2024-05-17T15:42:55.705049839Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 17 2024-05-17T15:42:55.705074209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 21 2024-05-17T15:42:55.705081209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 22 2024-05-17T15:42:55.705085879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 20 2024-05-17T15:42:55.705090319Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 18 2024-05-17T15:42:55.705094649Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 19 2024-05-17T15:42:55.705098879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 16 2024-05-17T15:42:55.705207399Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 15 2024-05-17T15:42:55.705459439Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 27 Without tcp_reuse_ports=yes, the segfault was always preceded by the following line if any existing TLS connections were stuck in TIME_WAIT: 2024-05-16T19:18:51.654447639Z 9(14) WARNING: {1 1 INVITE XXX(a)0.0.0.0} <core> [core/tcp_main.c:1301]: find_listening_sock_info(): binding to source address 10.X.X>X:5061 failed: Address already in use [98] 2024-05-16T19:18:51.746994728Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 14 exited by a signal 11 When the server wasn't handling any traffic, the issue didn't occur even in 5.7.4. Does anyone have any insights or suggestions on how to address this issue? Kind regards Stefan

2 2

Assistance Needed to Resolve Duplicate Contact Headers in 200 OK Response
by satyaprakash ch 17 May '24

17 May '24

Hi We are receiving 2 to 3 contact headers in the 200 OK response of the registration request. It appears that the `user_location` is updating multiple contact headers for the same extension due to network disconnections. Please suggest a solution to avoid duplicates in the `user_location` module, or how to ensure that only the latest single contact header is included in the 200 OK response. Your suggestion would be appreciated. Thanks and regards, Satyaprakash.

2 1

Fwd: rtpengine timestamp jumps
by Jeff Brower 17 May '24

17 May '24

Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message -----

1 0

$dlg_var() not implemented in ims_dialog module
by Valentin Christoph 17 May '24

17 May '24

Dear all, I would probably need to store some dialog scoped information in an IMS environment, but the ims_dialog module does not implement the $dlg_var() pseudo variable. Do you know other, maybe better, possibilities to store dialog scoped information in an IMS (CSCF) environment? Thank, Christoph

3 2

making HTTP Async request in stateless configuration
by Sergio Charrua 17 May '24

17 May '24

Hi all! Just wondering what would be you opinion on the following. We are integrating Kamailio as a stateless redirect server, using SIP 300 Multiple Choices response. The script requires that prior to the redirection, a HTTP request must be made. We thought about using HTTP Async Client to make the requests (POST of JSON content), which is our prefered way of sending HTTP requests, and depending on the response from the REST API, reply back to UAC a SIP 300 Multiple Choices message and terminate the call flow on Kamailio. From what I know, HTTP Async Client requires that the TM module be used, which (it is my understanding) will imply that the whole Kamailio script to be stateful, which I think is useless as the SIP flow will be really simple: INVITE -----> <------- 100 Trying <------- 300 Multiple Choices ACK ------> There is a catch, though, the CPS is very high, above 1000 CPS.... Could anyone share their thoughts on this? Should we go stateful and use the HTTP Async module? Should we go stateless and use the HTTP client module? What would be the best approach to implementing this? Thanks guys! *Sérgio Charrua*

1 0

WebRTC-UDP voicemail and call transfer
by Social Boh 16 May '24

16 May '24

Hello list, I'm trying implementing a solution like: https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg The 2 problems I have a this moment are: Call from UDP phone to WebRTC phone, the WebRTC phone reject the call; the call go to voicemail but in the SDP still have WebRTC SDP During a SIP UDP - WebRTC Call tranfer the call using phone button... the transfer don't work. Any hint Regards -- --- I'm SoCIaL, MayBe

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 14 May '24

14 May '24

How about this new function that sets the algorithm? -- Juha iff --git a/src/modules/auth/auth_mod.c b/src/modules/auth/auth_mod.c index 534ad9e20f..4e3a584d22 100644 --- a/src/modules/auth/auth_mod.c +++ b/src/modules/auth/auth_mod.c @@ -74,6 +74,10 @@ int w_consume_credentials(struct sip_msg *msg, char *s1, char *s2); * Check for credentials with given realm */ int w_has_credentials(struct sip_msg *msg, char *s1, char *s2); +/* + * Set authentication algorithm + */ +int w_auth_algorithm(struct sip_msg *msg, char *alg, char* s2); static int pv_proxy_authenticate( struct sip_msg *msg, char *realm, char *passwd, char *flags); @@ -170,6 +174,8 @@ static cmd_export_t cmds[] = { REQUEST_ROUTE}, {"pv_auth_check", (cmd_function)w_pv_auth_check, 4, fixup_pv_auth_check, 0, REQUEST_ROUTE}, + {"auth_algorithm", w_auth_algorithm, 1, fixup_spve_null, 0, + REQUEST_ROUTE}, {"bind_auth_s", (cmd_function)bind_auth_s, 0, 0, 0}, {0, 0, 0, 0, 0, 0} @@ -475,6 +481,35 @@ int w_has_credentials(sip_msg_t *msg, char *realm, char *s2) return -1; } return ki_has_credentials(msg, &srealm); + +} +/** + * + */ +int w_auth_algorithm(sip_msg_t *msg, char* alg, char* s2) +{ + if (fixup_get_svalue(msg, (gparam_t*)alg, &auth_algorithm) < 0) { + LM_ERR("failed to get algorithm value\n"); + return -1; + } + + if (strcmp(auth_algorithm.s, "MD5") == 0) { + hash_hex_len = HASHHEXLEN; + calc_HA1 = calc_HA1_md5; + calc_response = calc_response_md5; + } + else if (strcmp(auth_algorithm.s, "SHA-256") == 0) { + hash_hex_len = HASHHEXLEN_SHA256; + calc_HA1 = calc_HA1_sha256; + calc_response = calc_response_sha256; + } + else { + LM_ERR("Invalid algorithm provided." + " Possible values are \"\", \"MD5\" or \"SHA-256\"\n"); + return -1; + } + + return 1; } #ifdef USE_NC

3 6

302 Message Handling
by tfayomi＠untangledtechnology.com 14 May '24

14 May '24

Hi All, I'm having trouble understanding/finding resources on how to handle a 302 message I receive. I'm working on making an inbound sbc with kamailio with clearip integration. I'm successfully sending to kamailio, but am having trouble retrieving the response from clearip in kamailio. For some background, the intended call flow should be this: first it should send from caller to kamailio (caller > kamailio), then over tcp kamailio sends to clearip (kamailio > clearip) [This is what is working as expected] Then I should be able to signal to kamailio to continue the call to the intended destination. When clearip sends a code back (in this case 302), I want to be able to see that code and send the caller to the correct destination. The problem I keep running into is that I can't seem to see the 302 code in a reply, branch or failure root or checking the status. I'm just lost on what I am misunderstanding and would appreciate if anyone could point me in the right direction on how to approach retrieving the 302 message. I'm using kamailio version 5.6.3 and debian 12.1 Thanks in advance, Temi

2 1

Prefix match in htable
by Antonio Reale 13 May '24

13 May '24

Hello, I'm testing Kamailio 5.x as a redirecting server based on calling number. The calling numbers are stored on a DB and loaded in htable at runtime. When the INVITE or REGISTER arrive Kamailio checks if there is a match on $fU in htable (KEY_NAME) and if there is one, it replies with 302 adding the Contact HF with $fU@$sht(cli=>$fU)(with the domain part found in KEY_VALUE): append_to_reply("<Contact: sip:$fU@$sht(cli=>$fU)>\r\n"); If I use the full number as KEY_NAME it works as expected, but how can I match on a prefix instead of the whole $fU? For example, If I receive a call form 0655123456 I want to match it also if I have the prefix 0655 in htable. Do you have some advice to achieve this goal with htable or in a similar way? Thank you. Best regards. -- Antonio Reale

1 0

Kamailio and RTPengine and no voice on calls
by palany 10 May '24

10 May '24

I have an AWS ec2 instance with Kamailio and rtpengine configured. I am having a challenge of no voice on calls. I have sip client on my mobile phone and soft phone on my laptop and the two are on different networks behind nat. Signaling is working fine but there is no media and when I check on on the SDP the call logs are showing private ips instead of public ips : o=- 3 2 IN IP4 192.168.1.33 and c=IN IP4 172.31.36.50. I understand this is a NAT issue but I am failing to see where is the challenge. My rtpengine seems to be working fine . Below is my Kamailio.cfg file link : https://docs.google.com/document/d/1MKZJGlQjxYbdtw6QcCm9_EmvVPOkMs-YXjlq1ZM2 mmY/edit?usp=drive_link PCAP file for the call: https://docs.google.com/document/d/12gofLYhN0-lYIZORngIeuaYxLudLjiNdZgY504Ei XFs/edit?usp=drive_link rtpengine.conf https://docs.google.com/document/d/1QPxey49UzuN3Wd8jp8jc0mFhh31rr1alIO-gl4gP JEI/edit?usp=drive_link Please help.

2 3

base64 decoding issue
by Sergio Charrua 10 May '24

10 May '24

Hi all! I have been dealing with STIR/SHAKEN for a few weeks now, and while doing some tests I have found an issue which I can't find the reason for. The script has the following route logic: route[HANDLE_STIRSHAKEN] { xlog("L_INFO", "HANDLE_STIRSHAKEN - STIR/SHAKEN Logic"); if ($hdrc(Identity) == 0 ){ xlog("L_INFO", "HANDLE_STIRSHAKEN - No Identity Header -> Skipping this logic"); return; } #Verify Call Identity xlog("L_INFO", "HANDLE_STIRSHAKEN - Identity Header: $hdr(Identity)"); $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 1: $var(header_1)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 2: $var(header_2)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsing!"); $var(hdr_1_json_payload) = $(var(header_1){s.decode.base64}); # Header 1 xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsed!"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Header 1 payload = $var(hdr_1_json_payload) "); $var(hdr_2_json_payload) = $(var(header_2){s.decode.base64}); # Header 2 jansson_get_field($var(hdr_2_json_payload), "attest", "$var(attest_value)"); jansson_get_field($var(hdr_2_json_payload), "ppt", "$var(ppt_value)"); jansson_get_field($var(hdr_2_json_payload), "typ", "$var(typ_value)"); jansson_get_field($var(hdr_2_json_payload), "x5u", "$var(x5u_value)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - $var(attest_value) $var(ppt_value) $var(typ_value) $var(x5u_value) "); } All it does is parse the Identity header received on the initial INVITE, decodes it (it is encoded in base64) and writes some JSON values on log. Nothing serious.... As the Identity header is divided into multiple parts, separated by a ".", I get the first 2 parts in 2 private variables, as per lines: $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); Logs show correctly that both variables are populated: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 1: eyJhbGciOiAiRVMyNTYiLCJwcHQiOiAic2hha2VuIiwidHlwIjogInBhc3Nwb3J0IiwieDV1IjogImh0dHBzOi8vY2VydGlmaWNh dGVzLmNsZWFyaXAuY29tL2IxNWQ3Y2M5LTBmMjYtNDZjMi04M2VhLWEzZTYzYTgyZWMzYS83Y2M0ZGI2OTVkMTNlZGFkYTRkMWY5ODYxYjliODBmZS5jcnQifQ May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 230 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 2: eyJhdHRlc3QiOiAiQSIsImRlc3QiOiB7InRuIjogWyIxNDA0NTI2NjA2MCJdfSwiaWF0IjogMTU0ODg1OTk4Miwib3JpZyI6IHsi dG4iOiAiMTgwMDEyMzQ1NjcifSwib3JpZ2lkIjogIjNhNDdjYTIzLWQ3YWItNDQ2Yi04MjFkLTMzZDVkZWVkYmVkNCJ9 However, when decoding both values: xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); only $var(header_2) gets decoded: May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 38 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 1:* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 182 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 2: {"attest": "A","dest": {"tn": ["14045266060"]},"iat": 1548859982,"orig": {"tn": "18001234567"},"origid": "3a47ca23-d7ab-446b-821d-33d5deedbed4"}* As you can veri And to ease debugging, I have isolated the line that generates the issue: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsing! May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:351]: lval_pvar_assign(): non existing right pvar* *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:411]: lval_assign(): assignment failed at pos: (21,32-21,64)* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 27 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsed! Even though the $var(header_1) contains a base64 encoded string, the decoder seems to fail. If I copy the value contained in $var(header_1) to an online decoder like https://www.base64decode.org/ the string gets decoded correctly. I have even modified the line : $var(header_1)= $(hdr(Identity){s.select,0,.}); to get the reverse index from the Identity string by doing $var(header_1)= $(hdr(Identity){s.select,-6,.}); It gets the same encoded string, same value (as expected), but the result is the same: Kamailio's s.decode.base64 fails with the same error. i'm not sure what assignment failed at pos: (21,32-21,64) could mean.... I don't think my code is wrong, as the same function is used on $var(header_2) value and it works, but maybe someone can correct me. Greatly appreciate your help. Cheers, *Sérgio Charrua*

1 1

Failover with dialog dmq while dialog state 2
by Björn Klasen 09 May '24

09 May '24

Hello @all as we want to enhance our VoIP-Plattform with DMQ functionalities for replication of Userlocation, Dialog and Hash Tables I made a lot of tests durings the last week. For testing I use two Debian 12 Server with Kamailio 5.7.4 (latest git) Both server have their own floating IP so in case one server fails the floating changes to the working server. This is done via keepalived On both servers Kamailio is active and use ip_free_bind=1 so both Kamailio instances can listen to all necessary IPs. So we have following network configuration: Server1: static IP: 192.168.253.201 floating IP: 192.168.253.200 Server2: static IP: 192.168.253.202 floating IP: 192.168.253.210 Now I test some calls while I trigger a failover, so Server1 takes over 192.168.253.10 Message Test Result CustomerA -> Proxy -> CustomerB Failover while Dialogstate 4 BYE is relayed correctly. Dialog is changing to state 5 CDR needs to be written manually -> no problem at all CustomerA -> Proxy -> CustomerB Failover while Dialogstate 2 200 OK is relayed correctly although no transaction exist an therefore no t_reply route is triggered. This is not a big deal because i can use default onreply_route so all necessary variables can be used via hash table. BUT Big Problem: Dialog state stays 2, so a following ACK is a bogus event and the dialog state is never changed to 4. Even worse: after 300 seconds the dialog is cleaned away by the timer, so I have no chance to do any kind of accounting. Test 2 is a real mess, because for me there is only one solution. I have to check after a failover on incoming replies to an INVITE that was handled on Server2 if these current dialog state is < 4 and cancel them. But I can't see how to do it? Are there other ways than using t_cancel? Or is it even some kind of a bug that a 200 OK does not trigger state 3 on dialog (although I have in mind that it has to be something to do with missing transaction data, that are not replicated because of missing replication capabilities of tm module)? If none of the above is possible I really asking myself of a real use case of dialog dmq replication. -- Björn Klasen, Teamleitung NGN VoIP-Backbone TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 1

How to set "set_dialog_profile" value for Leg-a and leg-b.
by satyaprakash ch 09 May '24

09 May '24

Hi, We are using kamailio and freeswitch, So when we make a call from A to B the call is going via "kamailio -> freeswitch" and "freeswitch -> kamailio". Please suggest me how i can set "set_dialog_profile" for both legs with unique value in kamailio config file. Thanks.

1 0

Is there any way to modify the Refer-To header username?
by Shah Hussain Khattak 09 May '24

09 May '24

Hello Experts, I wanted to check, is there any way to modify the username part of the Refer-To header in Kamailio? For example, I am getting the following header: REFER-TO: <sip:sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> the username part is missing. Can I add a username inside it? For example, change it to REFER-TO: <sip:123@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> REFER-TO: <sip:test-kam@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> I could not find any solution in the mailing list, the only way was to remove the header and then re-add it. Thank you! Regards, Shah Hussain

2 2

rtpengine timestamp jumps
by Jeff Brower 09 May '24

09 May '24

Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff

1 0

Kamailio v5.7.5 Released
by Daniel-Constantin Mierla 08 May '24

08 May '24

Hello, Kamailio SIP Server v5.7.5 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.5. For more details about version 5.7.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/05/kamailio-v5-7-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.1 was released on April 3, 2024. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

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