sr-users January 2016

sr-users@lists.kamailio.org

86 participants
112 discussions

rtpengine to do DTLS-SRTP handshake, but media should not pass through it. the media should go through media server, which is running on the same machine
by riko nir 19 Jan '16

19 Jan '16

Hi, We have a media server and (opensips + rtpengine). How can I configure opensips+rtpengine to just do DTLS handshake with the client, but the media should not go through RTPEngine. The media should go to the Media server. Thanks. Riko

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Kamailio Load Balancing Cluster
by sandeep goje 19 Jan '16

19 Jan '16

Hi, I am using Kamailio as a load balancer with call load distribution to Freeswitch Gateways. I need a HA cluster of Kamailio. I have seen instructions to setup Kamailio HA cluster using KeepAlive but I also want them to share the load balancing information. I have seen OpenSIPS supports this. Is there a way in Kamailio to make both load balancers also share the load balancing information. Thank you. Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston

2 1

Get value of RPID privacy param
by Grant Bagdasarian 18 Jan '16

18 Jan '16

Hello, How do I select the value of the privacy parameter in the Remote-Party-ID header? Remote-Party-ID: "John Doe" <sip:jdoe@foo.com>;party=calling;id;privacy=full;screen=yes The following doesn't work, and I believe because it's not a valid select class: $sel((a)hf_value.remote_party_id.privacy) I'm using kamailio 4.1. Regards, Grant

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IMS terminating dialog does not re-run through all iFCs after calling t_newtran()
by Russell McConnachie 17 Jan '16

17 Jan '16

Hi, My CSCF environment has an application server acting as a B2BUA. It's originating a new session, and creating a new terminating session back to the S-CSCF (two separate sessions per-call). When I perform a re-INVITE with a diversion to transfer the call to another URI it is all good until the S-CSCF attempts to call the call control logic: # Start new transaction: t_newtran(); if (isc_match_filter("orig", "location")) { t_on_failure("ISC_ORIG_FAILURE"); exit; The t_newtran() method from what I understand will create a new transaction. Inside of this new transaction, I would think that the isc_match_filter should re-run through all of the iFCs defined on the HSS service profile, but it does not seem to be doing so. It seems to be continuing at where the original transaction ISC mark skip point was set. Thanks

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Different rtpproxy for different media type
by Koray Vatansever 15 Jan '16

15 Jan '16

Hi everybody, Is there a way to use different rtpproxies for different media types? I want to use one rtpproxy set for audio, and use another one for video. Is this possible in kamailio? Regards, Koray

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Kamailio replies with SIP/2.0 500 Service Unavailable Under load
by sandeep goje 15 Jan '16

15 Jan '16

Hi, I am doing load testing with of Kamailio. I have sipp (version 3.3) as the load generator,Kamailio(4.2.7) as Load Balancer and Freeswitch (1.6.5) as Gateways. I am using the following command to generate load ./sipp -sf uac.xml -d 2000 -m 4000 -l 1000 -r 200 -trace_stat -i SIPP_IP -p 15060 --trace_err -trace_error_codes -trace_calldebug -trace_screen KAM_IP When the rate of generation is below 80 (-r 80), Kamailio behaves fine.But if we go beyond 80, Kamailio starts sending "500 Service Unavailable" 2016-01-12 11:18:15.375103 1452577695.375103: Aborting call on unexpected message for Call-Id '96-32493@SIPP_IP': while expecting '180' (index 2), received 'SIP/2.0 500 Service Unavailable^M Via: SIP/2.0/UDP SIPP_IP:15060;branch=z9hG4bK-32493-96-0^M From: sipp <sip:sipp@SIPP_IP:15060>;tag=96^M To: sut <sip:service@KAM_IP :5060>;tag=55f576f507e822fa6633cf4bc22740e6-6660^M Call-ID: 96-32493@SIPP_IP^M CSeq: 1 INVITE^M Server: kamailio (4.2.7 (x86_64/linux))^M Content-Length: 0^M With the above command, sipp output is ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2016-01-12 11:26:07.723764 1452578167.723764 Last Reset Time | 2016-01-12 11:27:16.280901 1452578236.280901 Current Time | 2016-01-12 11:27:16.282737 1452578236.282737 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:001000 | 00:01:08:558000 Call Rate | 0.000 cps | 58.345 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 4000 Total Call created | | 4000 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 1192 Failed call | 0 | 2808 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000000 | 00:00:15:102000 Call Length | 00:00:00:000000 | 00:00:10:614000 ------------------------------ Test Terminated -------------------------------- Kamailio is run with command /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 8 -u kamailio -g kamailio version: kamailio 4.2.7 (x86_64/linux) 727746 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 727746 compiled on 15:59:03 Jan 11 2016 with gcc 4.6.3 System : Ubuntu 12.04.4 LTS Linux devops10-60-20-169 3.11.0-15-generic #25~precise1-Ubuntu SMP Thu Jan 30 17:39:31 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux I am not seeing any errors in Kamailio Logs. Any pointers on how to get it to serve more calls/sec Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston

4 3

Re: [SR-Users] Parser error when enabling siptrace with HEP functionality
by Sergey Basov 15 Jan '16

15 Jan '16

And one more thing. In captured packet from Homer, I see original SIP packets, without topoh modifications. But in dump from network interface by wireshark i see that packets was modified... -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com SATB-RIPE SATB-UANIC tel: (+38067) 403-62-54

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rtpengine graphite statistics server
by Juha Heinanen 15 Jan '16

15 Jan '16

rtpengine README.md mentions "graphite statistics server". is that server openly available somewhere? -- juha

3 3

Kamailio and NAT
by Nelson Migliaro 15 Jan '16

15 Jan '16

Hello, I am running Kamailio behind NAT. Kanailio has a private IP and I am relaying NAT to internet router. I am using: - #!define WITH_NAT - listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060 - Patched RTP proxy including the advertise option And everything goes fine. I can make calls and have two way audio. The problem begins when the callee ends the call. BYE is not received in Kamailio (caller) I included the public IP using "add_contact_alias" because "set_contact_alias" was not adding the public IP. I included this in in NATDETECT (pre loaded router) if(is_first_hop()) { xlog("L_NOTICE","Metodo: $rm \n"); xlog("L_NOTICE","is first hop\n"); #set_contact_alias(); if (!add_contact_alias("PUBLIC-IP", "$Rp", "udp")) { xlog("L_ERR", "Error in aliasing contact $ct\n"); send_reply("400", "Bad request"); exit; } } I think the problem is related to destination that BYE is sent by the vendor. From what I see IP and port is taken from advertised in contact (PUBLIC-IP and 5060). The problem is that internet router changes the source port. Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1> --- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp", "udp") in order to received new transactions or should I follow a different procedure??? Thank you

4 14

dispatcher state x and the return of the probing mode
by Sven Neuhaus 14 Jan '16

14 Jan '16

Hello, when trying to set a destination to inactive using dispatcher.set_state after setting the state to "i", when I do a "dispatcher.list" I see the state as "ix". The "x" state appears not to be documented. After a short time (seconds) the flags return to state "ip" (probing). According to the module documentation I should be able to set the "probing" mode. 1. Why doesn't probing mode remain disabled? 2. What is state "x"? Cheers, -Sven

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sr-users January 2016