sr-users September 2024

sr-users@lists.kamailio.org

37 participants
52 discussions

Handling SIP Contact Header Changes and ACK Routing Issues
by Josimar Rocha 17 Sep '24

17 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: Asterisk (internal) → Kamailio (Proxy) → SIP Provider (PSTN) Both Asterisk and Kamailio are under my responsibility. Problem: When I initiate a call from Asterisk, I use Kamailio as the outbound proxy. The INVITE is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a 200 OK, indicating the call was answered. However, the 200 OK received from the provider contains a modification in the Contact header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent ACK should be sent. The issue arises when Kamailio receives this 200 OK: it overwrites the Contact field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the ACK back to Kamailio, which, instead of forwarding it to the new IP from the Contact in the 200 OK, sends it to the SIP provider at the same address used for the initial INVITE. Since the ACK does not reach the IP expected by the provider (the new IP introduced in the Contact), the provider continues to send repeated 200 OK responses, until the call is disconnected due to a timeout, as a valid ACK was never received. I looked for conceptual answers in RFC 3261 to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. Key Questions: Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK? Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE? How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?

1 0

Handling SIP Contact Header Changes and ACK Routing Issues
by Josimar Rocha 17 Sep '24

17 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: Asterisk (internal) → Kamailio (Proxy) → SIP Provider (PSTN) Both Asterisk and Kamailio are under my responsibility. Problem: When I initiate a call from Asterisk, I use Kamailio as the outbound proxy. The INVITE is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a 200 OK, indicating the call was answered. However, the 200 OK received from the provider contains a modification in the Contact header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent ACK should be sent. The issue arises when Kamailio receives this 200 OK: it overwrites the Contact field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the ACK back to Kamailio, which, instead of forwarding it to the new IP from the Contact in the 200 OK, sends it to the SIP provider at the same address used for the initial INVITE. Since the ACK does not reach the IP expected by the provider (the new IP introduced in the Contact), the provider continues to send repeated 200 OK responses, until the call is disconnected due to a timeout, as a valid ACK was never received. I looked for conceptual answers in RFC 3261 to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. Key Questions: *Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK? *Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE? *How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?

1 0

About error running PCSCF
by luis.ramost＠pucp.edu.pe 16 Sep '24

16 Sep '24

Hi community: I am trying to run the Kamailio IMS, but I have the next error in the module PCSCF I installed the Kamailio on a VM Machine, with UBuntu 18.04 Commands: $ sudo -i # mkdir -p /var/run/kamailio_pcscf #kamailio -f /etc/kamailio_pcscf/kamailio_pcscf.cfg -P /kamailio_pcscf.pid -DD -E -e Output: loading modules under config path: /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/modules 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <delete_delay> of type <2:int> not found in module <ims_registrar_pcscf> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 399, column 51: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <ipsec_preferred_ealg> of type <1:string> not found in module <ims_ipsec_pcscf> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 417, column 59: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <recv_mode> of type <2:int> not found in module <ims_qos> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 435, column 35: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <dialog_direction> of type <2:int> not found in module <ims_qos> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 436, column 42: Can't set module parameter 0(470) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/register.cfg, line 260, column 42: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_create (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/register.cfg, line 264, column 77: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mo.cfg, line 114, column 31: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mo.cfg, line 180, column 33: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mt.cfg, line 11, column 34: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mt.cfg, line 99, column 42: unknown command, missing loadmodule? ERROR: bad config file (10 errors) 0(470) WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) 0(470) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... 0(470) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): Error destroying spi generator 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:313]: mod_destroy(): Error destroying port generator I guided form this tutorial: https://ryantheelder.github.io/blog/2023/VoLTE/ Please if somebody would help me, I appreciate it Finally my objective is to connect with srsRAN and open5GS to have VoLTE service. Best wishes, Luis

4 3

[Dialog module] Dialog Flag Removal and recommendation of dlg_manage()
by Xenofon Karamanos 16 Sep '24

16 Sep '24

Hello all, We would like to inform you about a recent PR<https://github.com/kamailio/kamailio/pull/3961> (https://github.com/kamailio/kamailio/pull/3961) that introduces changes related to the dialog flag handling. Summary of Changes: * Removal of dlg_flag: The dlg_flag has been removed from the dialog module. * Replace with call of dlg_manage(): We recommend using the function dlg_manage() as a replacement to handle the necessary functionality. Please refer to the updated documentation for more details. Affected Modules: 1. These modules only required updated documentation that references the use of dlg_flag instead dlg_manage(). * acc * qos * pua_dialoginfo * sst 1. Usage Changes (Dialog Flag Handling Removed): In these modules, the code related to reading and checking the dialog flag has been removed. Additionally, the setting of the message internal flag has been removed. If you need to track the dialog or call state, you MUST now call dlg_manage() explicitly in the kamailio config * call_control * cnxcc * mediaproxy * nat_traversal Please review the updated documentation for further guidance on the use of dlg_manage(). Feel free to reach out with any questions, concerns and report any issue due to this change. Best regards, Xenofon Karamanos

1 0

rtpengine module
by Daniel Donoghue 16 Sep '24

16 Sep '24

Hello, I am using rtpengine to encrypt and decrypt DTLS and SDES audio streams for my calls. I have a set of rtpengine instances running, configured using a database, such that it is easy to scale up on demand. This works well for initiating a call and balancing load across multiple instances but I want to be able to recover from rtpengine failure. I have looked at configuring rtpengine with a redis service but do not believe that to be helpful in this case as im using encrypted media and because the configuration of rtpengine with redis has a lot of overheads, limits scaling and requires non standard configuration of a redis cluster. I would like to be able to re-anchor media streams instead of relying on rtpengine to synchronise itself across a redis cluster. For this to work, i believe i would have to use the uac module to send a reINVITE and I speculate that i would need to use an xhttp event to trigger this remotely when a failure is detected. I am unclear, though, how to get to a specific call within kamailio from the xhttp route in order to do anything to it, if i even need to, and i am nuclear how to make the rtpengine module select a new node for sdp for the reinvite? How can i achieve my goal? Perhaps it would be possible to modify the rtpengine module itself to support generating a reinvite and selecting a new node on receipt of an rpc request? Hoping someone can help me find a way forward With every blessing, Daniel

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About Rx AAR in PCSCF for mobile terminated call
by Liu Wing Hong 16 Sep '24

16 Sep '24

Hi, I am using kamailio 5.8.3 as an IMS-PCSCF. On handling SIP INVITE (and response), it sends Rx AAR to PCRF for bearer establishment. For mobile-terminated call, it was found that the call was dropped by handset within several seconds after ringing for some handsets. The corresponding pcap capture is attached for reference. From the pcap capture, it was found that in the Rx AAR request sent (packet 11), in the Media-Sub-Component Flow-Description, the "permit out from ip/port" and "permit in to ip/port" were set to 202.0.169.182 port 24428 instead of 10.51.0.15 port 31124. 202.0.169.182 port 24428 is the one in the incoming INVITE sdp from S-CSCF (packet 2); 10.51.0.15 port 31124 is the one in the processed INVITE sdp sent to UE (packet 5); Also, the sdp data of "offer" inside AAR was set to the sdp in the incoming INVITE from S-CSCF instead of the sdp in the processed INVITE sent to UE. Is that correct? Please advice. Thank you. Regards, Hong

1 0

Setting $du to websocket-secure
by James Browne 13 Sep '24

13 Sep '24

How can I set the destination URI for an INVITE to be a websocket-secure destination? Is it possible? Summary I've a proxy with tcp_connection_match=1, but websocket URIs always have transport=ws (never transport=wss) in them, so relaying a call to a WSS connection always fails. I tested running kamailio 6.0.0-dev2 compiled from a commit made this week. This proxy server uses nathelper rather than outbound module. Detail We know that "transport=ws" is used for both WS and WSS. I've a proxy server that receives an INVITE for a WSS destination, and this proxy supports both WS and WSS. This proxy server must have core parameter tcp_connection_match=1 set, and this leads the t_relay() to fail. When an INVITE comes, these are the steps. - The URI is something like sip:user@anonymous.invalid;alias=198.51.100.10~52833~6;transport=ws. - First handle_ruri_alias() removes the alias (which has ~6 in it, for wss) and sets the $du to something like sip:198.51.100.10:52833;transport=ws. - Then loose_route_preloaded() processes the Route header fields and forces the outbound socket to the TLS websocket one. - Then t_relay() fails to relay the INVITE and responds with 477 or 500. If, however, there's a non-TLS websocket connection open to the proxy, the INVITE would be erroneously relayed over that (using the wrong kamailio-side TCP port). I can go deeper with testing if required. I wonder whether this is a bug. James

3 12

Strange result: $stat(total_size) returns 0
by whliu＠citictel.com 12 Sep '24

12 Sep '24

Hi, I am using kamailio 5.8.3. I just found that strange result is returned from variable $stat(total_size). In the example script under "misc/examples/ims/pcscf/kamailio.cfg", there is a line, $var(used) = 1 - ($stat(free_size) / $stat(total_size)); The trace log said that $stat(total_size) was returning 0, and got a log like this ERROR: <core> [core/rvalue.c:1434]: long_longop2(): rv div by 0 Any idea on this? Please advise. Thank you. Regards, Hong

1 0

Strange result: $stat(total_size) returns 0
by whliu＠citictel.com 12 Sep '24

12 Sep '24

1 0

SIP parse error for self-generated Websocket reply
by Sebastian Damm 12 Sep '24

12 Sep '24

Hello, We try to introduce a small websocket sidecar to our Kamailio servers, which runs on the same host. It connects to the regular TCP socket of our Kamailios, and in the XHTTP request route, we do just this: if path == "/sidecar": KSR.websocket.handle_handshake() return 1 On every new connect, Kamailio logs a parse error for the reply it generates. Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 22) Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols Sia: SIP/2.0/TCP 100.74.11.1:46422 Sec-WebSocket-Protocol: sip Upgrade: websocket Connection: upgrade Sec-WebSocket-Accept: 60dPbzcRW6up7z92kDlxkqL2WlY= Content-Length: 0 > Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/msg_translator.c:3402]: build_sip_msg_from_buf(): parsing failed I saw this error in other questions from a while ago, but without a solution. What could be the reason for this? Thanks in advance, Sebastian

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sr-users September 2024