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37 participants
37510 discussions

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

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

08 May '24

Hello, I am considering to release Kamailio v5.7.5 (out of branch 5.7) on Wednesday, May 8, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

sql_query fails from event_route
by Juha Heinanen 07 May '24

07 May '24

I noticed that sql_query does not (anymore) work when called from event route: event_route [tcp:closed] { # Handle TCP connection close # Delete possible leftover registration if ($conid) sql_query("sip_proxy_cache", "DELETE FROM location WHERE server_id=0 AND connection_id=$conid", "result"); } The query produces errors: 2024-05-07T11:30:14.785915+03:00 siika /usr/bin/sip-proxy[2848078]: CRITICAL: sqlops [sqlops.c:213]: sql_check_connection(): no database handle with reconnect disabled 2024-05-07T11:30:14.785978+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:232]: sql_query(): invalid connection to database 2024-05-07T11:30:14.786013+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:237]: sql_query(): with query [DELETE FROM location WHERE server_id=0 AND connection_id=7] I added the last debug message in order to find out where the error comes from. The same query works fine when executed from regular route. Any ideas why the query fails from event_route? -- Juha

2 3

How to use environment variable
by Pavan Kumar 07 May '24

07 May '24

Hi everyone, I am trying to assign environment variable as follows listen=udp:0.0.0.0:5060 advertise $env(MY_IP):5060 Looks like using the environment variable as above is an invalid configuration. Is there a way to use IP from env var to advertise. Even better, is there a way to use result in a stun query as an advertised address? Thank you, Pavan Kumar

2 3

Retransmission behaviour after ACK
by Sergio Charrua 06 May '24

06 May '24

Hi all! I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing. The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules. Request route is simple: request_route { route(HANDLE_OPTIONS); route(HANDLE_RETRANS); route(HANDLE_CANCEL); route(REQINIT); route(HANDLE_INVITE); } where HANDLE_RETRANS is: route[HANDLE_RETRANS]{ # handle retransmissions xlog("L_INFO","HANDLE_RETRANS - message type is $mt for method $rm"); if (!is_method("ACK")) { if(t_precheck_trans()) { xlog("L_INFO","HANDLE_RETRANS - Precheck Trans"); t_check_trans(); xlog("L_INFO", "HANDLE_RETRANS - Exiting after Retransmission check - method $rm"); exit; } xlog("L_INFO","HANDLE_RETRANS - Check Trans"); t_check_trans(); } } and HANDLE INVITE is: route[HANDLE_INVITE]{ if (is_method("INVITE")) { xlog("L_INFO","MAIN - calling TO_CARRIER"); setflag(FLT_ACC); route(TO_CARRIER); exit; } } The HTTP requests call a REST API returning a JSON object, that is iterated and produces a CONTACT header containing multiple choices using the append_branch() command. The RELAY route is : route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","REPLY_302 - send reply"); send_reply("300", "Multiple Choices"); xlog("L_INFO","REPLY_302 - exit"); exit; } Which returns indeed a Contact header with multiple contact choices. When using Asterisk to originate a call to Kamailio, the SIP call flow is as follows and terminates (SNGrep and TCPDump do not detect anything else): SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> However when using SIPp I get some retransmissions after the final ACK. SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── SIPp scenario is : <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC">   <send retrans="500" start_rtd="true" rrs="true"> <![CDATA[ INVITE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]> Contact: sip:sipp@[local_ip]:[local_port];transport=[transport]> Call-ID: [call_id] CSeq: 102 INVITE User-Agent: SIPp Supported: replaces, timer Content-Type: application/sdp Subject: Performance Test Session-Expires: 3600;refresher=uas Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIPp c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" rss="true"> </recv> <recv response="300" rtd="true" rss="true"> <action> <ereg regexp="\<([a-zA-Z0-9@.=;]+)[^>]+" search_in="hdr" header="Contact: " assign_to="1"/> <log message="INFO: 300 Contact Header: [$1]" /> <exec command="echo [$1] >> ./300_ContactHeader.txt" /> </action> <action> <ereg regexp="sip:.*@([0-9A-Za-z\.]+):([0-9]+);transport=([A-Z]+)" search_in="hdr" header="Contact:" check_it="true" assign_to="dummy,host,port,transport" /> </action> </recv>   <send rrs="true"> <![CDATA[ ACK sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]>[peer_tag_param] Contact: <sip:sipp@[local_ip]:[local_port]> Call-ID: [call_id] CSeq: 102 ACK User-Agent: SIPp Content-Length: 0 ]]> </send>   </scenario> The question is : is there anything wrong with my kamailio script? or with SIPp scenario? why does kamailio have different behaviours between Asterisk and SIPp? Any option to disable retransmissions on Kamailio ? (using UDP) Thanks in advance, *Sérgio Charrua*

2 3

topos with mask_callid
by Marrold 05 May '24

05 May '24

Hi all, I am using Kamailio 5.8.1 on a Debian 12 machine with topos and topoh hiding the topology. I have observed with modparam("topos", "mask_callid", 1), in-dialog requests are relayed with the incorrect caller-id. In the scenario, A calls B. The call is set up and the call-id for the B-Leg is successfully masked. Then, when A sends a BYE, the BYE is incorrectly relayed to B with the call-id of the A leg, and B replies with a 481 If I disable masking the call-id, or disable topos/topoh entirely, everything looks and works as it should. I have tested with 5.7.4 and the behaviour is the same. Does anyone have any ideas for where I should start looking? Thanks Matthew

1 0

Re: IMS SCSCF redundancy
by Oleg Belousov 03 May '24

03 May '24

Hi. I have a question re: IMS SCSCF redundancy.:There are several SCSCF instances, registration happens via SCSCF1, it goes down after that. UE make MO call.. PCSCF resolves the common SCSCF fqdn and sends an invite to available SCSCF2 which does not have UE profile as well as AS address (which is received during registration SAR/SAA in iFC). There are 2 options here, either to release the call (as per 3gpp spec upon specific release code, eg 408/503 UE should perform re-registration - not happen in our case unfortunately), or to send request for iFC (suspend the call, fire SAR, receive SAA, check available services, etc) and upon success continue the call (to AS first)/ Lke IMS Registrar SCSCF save function, but for the invite. To continue the call without interaction with AS is not a good idea.. Any advice or workaround on how to achieve desired SCSCF redundancy ? -- obelousov.tel

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