- sr-users - kamailio.org

Branching and rejects.
by Kjeld Flarup 18 Apr '18

18 Apr '18

Hello What is the correct way to handle, if a branched call reaches a 404 on one of the branches. Currently I discard all 404 in onreply_route, but if all branches has 404, then this is never send to the caller. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 2

Failure Route Issue
by mack＠dopensource.com 18 Apr '18

18 Apr '18

Hey All, I have a weird issue. I’m sending a SIP INVITE to a SIP Service that send’s me back a 433, which let’s me know that the number is invalid. I then send an INVITE to a media server (FreeSwitch) to play a message that the number can’t be called, which looks to be working. But, the final ACK from the SIP client is sent back to the SIP Service versus the media server. The following diagram below depicts this: https://snag.gy/qMWfvk.jpg. The code snippet in the failure_route for the the SIP Service has this: if (t_check_status("403|433")) { if ($avp(s:teleblock_media_enabled) == "1") { # make sure media server can route back to kamailio record_route(); revert_uri(); $ru = "sip:" + $T_reply_code + $rU + "@" + $sel(cfg_get.teleblock.media_ip) + ":" + $sel(cfg_get.teleblock.media_port); append_branch(); t_relay(); exit; }

2 5

Kamailio Websocket Questions
by Nagorny, Dimitry 17 Apr '18

17 Apr '18

Hi List, two questions regarding Kamailio Websockets: 1. How do I send INVITE request TO the listening Websocket? I usually used for UDP/TCP/TLS these format, but somehow this is not working for Websocket because it gets in an endless loop: if ($rU=~"^(\+|00|0)?[1-9][0-9]{4,20}$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ";user=phone;transport=udp"; force_send_socket(udp:MY_EXTERN_IP:MY_EXTERN_PORT); rtpengine_offer("internal trust-address RTP AVP"); route(RELAY); exit; } 2. Why do I get these errors every time my Websocket Client is registering, but it still works and DB is inserted with registered location: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TCP 10.250.5.17:51283#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: 6uLXlD/aBrzu/j3PYP3DWO7rvLE=#015#012Server: kamailio (4.3.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> I am using Kamailio 4.3.5 + CentOS 7.2. I appreciate every help I can get. Best Regards Dimitry Nagorny Trainee

3 4

Dialog module's keep-alive feature uses wrong CSeq?
by Oded Arbel 17 Apr '18

17 Apr '18

Hi all. I'm trying to use the dialog module's keep alive feature (ka_interval() and dlg_set_property()), and the documentation (under dlg_set_property) says this: If keep alive is enabled for a dialog, the module will send SIP OPTIONS requests with CSeq lower or equal than last request within dialog This is seems really weird to me, but the problem is worse than that - when setting dlg_set_property("ka-src"), the OPTIONS messages sent to the caller always have CSeq set to "0", regardless of what CSeq the UA sent on its invite. The problem is that while some UAs respond to these purposefully broken messages with some kind of a 5xx reply (for example, Linphone funnily sends "500 Internal Server Error"), several UAs don't respond at all. Worse than that - some of the UAs that send back 5xx replies to OPTIONS with a non-zero but invalid CSeq will completely ignore the OPTIONS keep-alive message that is sent to the caller with a CSeq 0 - for example, PJSIP responds with "503 Invalid CSeq" to a "ka-dst" OPTIONS message but will ignore a "ka-src" OPTIONS message with CSeq 0. My questions: 1. Why are keep-alive messages sent with a wrong CSeq? 2. If we're keeping the "wrong CSeq" approach, can we at least not send CSeq "0"? 3. Is there a way to change any of the above behaviors with a configuration (i.e. without patching the sources)? Thanks in advance -- Oded Arbel odeda(a)cloudonix.io

3 5

Kamailio with 2 Public and Private IPs- SIP Call flow issues
by arik halperin 16 Apr '18

16 Apr '18

Hello, I have the following setup: Kamailio: Public Interface 70.36.25.65 Private Interface 10.168.10.227 Freeswitch: Private Interface 10.168.10.200 I’m routing calls from kamailio to freeswitch, but when I try to hang up the call the freeswitch sends the BYE to the public address(70.36.25.65) and not to the private address 10.168.10.227. 200OK and ACK work correctly. It’s only the BYE that has an issue. Showing INVITES: Client Invite: T 82.80.164.63:53871 -> 70.36.25.65:5099 [AP] INVITE sip:+972549428460@70.36.25.65:5099 SIP/2.0. Via: SIP/2.0/TCP 10.0.0.42:51619;branch=z9hG4bK-524287-1---52dcc73ec891de14;rport. Max-Forwards: 70. Contact: <sip:972545234585@82.80.164.63:51620;rinstance=1f9d21b54116df89;transport=tcp>. To: <sip:+972549428460@70.36.25.65:5099>. From: "972545234585"<sip:972545234585@70.36.25.65:5099>;tag=902c5325. Kamailio to FS: T 10.168.10.227:46506 -> 10.168.10.200:5080 [AP] INVITE sip:+972549428460@70.36.25.65:5099 SIP/2.0. Record-Route: <sip:70.36.25.65:5099;transport=tcp;lr=on;ftag=902c5325;nat=yes>. Via: SIP/2.0/TCP 70.36.25.65:5099;branch=z9hG4bK1117.1b91cfe0632bd6f036b308f85898e862.1;i=1. Via: SIP/2.0/TCP 10.0.0.42:51619;received=82.80.164.63;branch=z9hG4bK-524287-1---9b6758521ff0f528;rport=53871. Max-Forwards: 69.

3 3

Problem with resuming suspended transacation
by Arik Halperin 16 Apr '18

16 Apr '18

Hello, I’m trying to resume a suspended transaction, but keep getting this error: WARNING: tm [t_suspend.c:193]: t_continue(): transaction is not suspended [20608:1256194941] WARNING: tmx [tmx_mod.c:686]: w_t_continue(): resuming the processing of transaction [20608:1256194941] failed WARNING: tm [t_lookup.c:1483]: t_unref(): script writer didn't release transaction I’m implementing push handling, in INVITE I do the following: route[INVITE] { if (!lookup("location")) { send_reply("100", "Trying"); record_route(); route(SUSPEND); } else { # NAT detection route(NATMANAGE); if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); record_route(); t_relay(); ts_store(); $sht(vtp=>stored::$rU) = 1; } route(SENDPUSH); } # suspend the transaction route[SUSPEND] { if(!t_suspend()) //Transaction is suspended! { exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" +$T(id_label); } In Register I call the PUSHJOIN route: # append branches or resume the transaction route[PUSHJOIN] { $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $var(hstored) = $sht(vtp=>stored::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if ($var(hjoin)==0) { if ($var(hstored)) ts_append("location", "$tu"); return; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } For some reason I keep getting: Apr 12 12:37:11 kamprod /usr/local/sbin/kamailio[6039]: ERROR: <script>: 1111 suspended transaction [20608:1256194941] FreeSWITCH => 972587102881 pr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: ERROR: <script>: 1111 tU=972587102881 20608:1256194941 0 Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: ERROR: <script>: 1111 resuming trasaction [20608:1256194941] 972587102881 (20608:1256194941) Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tm [t_suspend.c:193]: t_continue(): transaction is not suspended [20608:1256194941] Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tmx [tmx_mod.c:686]: w_t_continue(): resuming the processing of transaction [20608:1256194941] failed Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tm [t_lookup.c:1483]: t_unref(): script writer didn't release transaction

3 2

Audio stops after resuming call from hold
by gerry kernan 16 Apr '18

16 Apr '18

Hi Hoping someone can point me in the right direction. I have a Kamailio Ver: 4.2.3-1.1 running in front of a few asterisk servers Ver: 13.17.2 sip is routed to an asterisk server depending the domain name in the sip request, all working as expected . but if a call is put on hold after resuming the call the party that placed the call on hold can't hear any audio. The other party can hear . do I need to do anything special to handle re-invites for calls put on hold? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

3 8

Help me!
by Do Quang Trung 15 Apr '18

15 Apr '18

I use kamailio 5.1.1 with openssl-1.0.2n. I built openssl with gost engine supported and the result when i list cipher list as follow: .................... GOST2001-GOST89-GOST89:GOST94-GOST89-GOST89:ECDH-RSA-AES256-GCM-SHA384:ECDH-ECDSA-AES256-GCM-SHA384:ECDH-RSA-AES256-SHA384:ECDH-ECDSA-AES256-SHA384:ECDH-RSA-AES256-SHA:ECDH-ECDSA-AES256-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA::GOST94-GOST89-GOST89:ECDH-RSA-AES256-GCM-SHA384:ECDH-ECDSA-AES256-GCM-SHA384:ECDH-RSA-AES256-SHA384:ECDH-ECDSA-AES256-SHA384:ECDH-RSA-AES256-SHA:ECDH-ECDSA-AES256-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA: ................ i must use exactly GOST2001-GOST89-GOST89 in kamailio to protect sip protocol. Please help me to resolve my problem. Best regards, Trung.

3 2

Kamailio 4.2.8 has crashed
by igor.potjevlesch＠gmail.com 13 Apr '18

13 Apr '18

Hello, I had a crash on one Kamailio instance with the following backtrace: Core was generated by `/usr/local/sbin/kamailio -m 704 -M 128 -P /run/kamailio/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00007f92366761f7 in raise () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 glibc-2.17-196.el7_4.2.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 libacl-2.2.51-12.el7.x86_64 libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 libcom_err-1.42.9-10.el7.x86_64 libdb-5.3.21-21.el7_4.x86_64 libgcc-4.8.5-16.el7_4.1.x86_64 libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7_4.1.x86_64 lm_sensors-libs-3.4.0-4.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 mariadb-libs-5.5.56-2.el7.x86_64 net-snmp-agent-libs-5.7.2-28.el7_4.1.x86_64 net-snmp-libs-5.7.2-28.el7_4.1.x86_64 nspr-4.13.1-1.0.el7_3.x86_64 nss-3.28.4-15.el7_4.x86_64 nss-softokn-freebl-3.28.3-8.el7_4.x86_64 nss-util-3.28.4-3.el7.x86_64 openssl-libs-1.0.2k-8.el7.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-292.el7.x86_64 popt-1.13-16.el7.x86_64 rpm-libs-4.11.3-25.el7.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 (gdb) backtrace #0 0x00007f92366761f7 in raise () from /lib64/libc.so.6 #1 0x00007f92366778e8 in abort () from /lib64/libc.so.6 #2 0x000000000061fb2c in qm_free (qm=0x7f91f4eaf000, p=0x7f92110d0300, file=0x7f9226dd7072 "dialog: dlg_hash.c", func=0x7f9226dd9aeb <__FUNCTION__.12761> "destroy_dlg", line=381) at mem/q_malloc.c:474 #3 0x00007f9226d6fb62 in destroy_dlg (dlg=0x7f920f775598) at dlg_hash.c:381 #4 0x00007f9226d75681 in dlg_unref (dlg=0x7f920f775598, cnt=2) at dlg_hash.c:874 #5 0x00007f9226da14d5 in dlg_ontimeout (tl=0x7f920f7755f8) at dlg_handlers.c:1488 #6 0x00007f9226db1e67 in dlg_timer_routine (ticks=8621904, attr=0x0) at dlg_timer.c:283 #7 0x00000000004af537 in compat_old_handler (ti=137950479, tl=0x7f91f7ad8cc8, data=0x7f91f7ad8cc8) at timer.c:1011 #8 0x00000000004aff14 in slow_timer_main () at timer.c:1145 #9 0x000000000052a2af in main_loop () at main.c:1684 #10 0x000000000052fe71 in main (argc=7, argv=0x7fff51f21418) at main.c:2581 (gdb) Is there enough information to understand the reason of the crash? Regards, Igor.

3 5

Help needed to compile and install kamailio 5.1.2
by vinay kumar 13 Apr '18

13 Apr '18

Hi All, can anyone share me the installtion steps for kamailio 5.1.2? can i compile and run kamailio 5.1.2 on ubuntu 14.04? i have already used 4.3.6 with polaris configuration to bringup kamailio and i wanna upgrade to kamailio 5.1.2 and the same way can i use 5.1.2 with polaris configuration? I have used polaris scripts to bringup kamailio 5.1.2 and its failing with status kamailio service with Result resources. if i use latest debian packages , are polaris configurations are compatible with latest packages?? Please help me out. Thanks in advance. Regards, vinay

1 0

Is RTPProxy the way to go?
by Robert Johnson 12 Apr '18

12 Apr '18

I'm encountering a situation where an endpoint will renegotiate it's RTP ports but RTPProxy does not honor the new ports and continues to send audio to the old ports. I realize that I'm running a (very) old version of RTPProxy and that I should upgrade before submitting any bugs or asking for help with the program. My question is, is RTPProxy still the way to go? And if so, is v2.0.0 the best version to go with? https://github.com/sippy/rtpproxy/releases At a basic level, the network looks like this +---------------+ +--------------+ | | | | | Priv. Network | | Pub. Network | | | | | +------+ +---------+--+ +---+-------+--+ +--+--------+ | | | | | | | | | PSTN <---> Softswitch <--------> Kamailio + <--------> EndPoints | | | | | | RTPProxy | | | +------+ +---------+--+ | | +--+--------+ | +---+-------+--+ | | | | | +---------------+ +--------------+

2 1

Kamailio + tls
by Do Quang Trung 12 Apr '18

12 Apr '18

Plz help me. I want config kamailio 5.1 support tls using gost89 cipher to protect data.

2 3

Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive delivery
by Abdul Basit 12 Apr '18

12 Apr '18

Hi Team, My issue for IM handling has been resolved. @MS helped to look into the matter. From to the example https://www.kamailio.org/docs/modules/5.1.x/modules/msilo.html#idp45433036 # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) I replaced it as below # -- tm params -- modparam("tm", "fr_timer", 10000 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) Lower fr_timer was initiating 408 without waiting for 200 OK from destination because 10ms is too low. This was confusing msilo module and the sender device that MESSAGE sent was failed. Increasing the fr_timer value resolved the issue. This also resolved a ripple effect. i.e, kamailio delivery of offline messages from DB store to the destination party. Since kamailio was get message delivery error, it was executing failure_route[1] that was storing the message again in DB store as offline message and so on. lower fr_timer values are also exits in 3.x, 4.x, 5x and dev branches documentation. Correction is required. -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > > On 9 April 2018 at 19:34, Abdul Basit <basit.engg(a)gmail.com> wrote: > >> Dear Team, >> >> I have added traces via pasetbin >> >> MESSAGE Stored When B is OffLine https://pastebin.com/NLwMePmf >> >> [trace screenshot removed] >> >> Next this messages should be delivered when B party REGISTER with the >> server >> >> *Issue 1*: Message was delivered to B-party but it didn't remove form DB >> and as a result MESSAGE deliver to B-party as many times it come line >> https://pastebin.com/7pb0wCKx >> [trace screenshot removed] >> >> Now next issue is >> *Issue 2*: Messages deliver to B-party when its already online without >> storing in DB. B-party receive MESSAGE successfully but A-party gets >> MESSAGE delivery failed. >> https://pastebin.com/zgDbK4F6 >> >> [trace screenshot removed] >> >> I intentionally modified the 202 - description for better troubleshooting. >> >> See if attachments were received properly through mailing list. >> >> For convenience I have attached the pcap as well. >> >> MESSAGE: 1: Accepted and Stored as expected >> MESSAGE: 2: Server delivered Message when B comes online >> MESSAGE: 3: Both A & B are online. A-party receives MESSAGE delivery >> failure where as B-party receives MESSAGE successfully >> >> >> Modified default kamailio.cfg is https://pastebin.com/thZig5RC >> It looks that mingled up the configuration. >> >> Please see whats gone wrong here. >> >> >> -- >> regards, >> >> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >> >> On 9 April 2018 at 18:20, Abdul Basit <basit.engg(a)gmail.com> wrote: >> >>> Dear team, >>> >>> Thank you for your reply. >>> >>> @MS & @ Davld, Let me test again and share traces. >>> >>> @Zaka, I will check the save() result as you suggested. Location is >>> being saved and properly looked up because messages are being delivered >>> (and not stored in DB ) when both parties are online. >>> >>> -- >>> regards, >>> >>> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >>> >>> On 9 April 2018 at 16:38, Muhammad Zaka <muhammad.zaka(a)cloudcall.com> >>> wrote: >>> >>>> Please check the return value of save and replace the logic: - >>>> >>>> >>>> >>>> $var(save_result) = save("location") ; >>>> >>>> if ($var(save_result) <= -1) { >>>> >>>> xlogl("L_ERROR", "[$rm] -> Unable to save user location \n"); >>>> sl_reply_error(); >>>> >>>> exit(); >>>> >>>> } >>>> >>>> >>>> >>>> >>>> >>>> if ($var(save_result) == 1) { >>>> >>>> xlogl("L_INFO", "[$rm] -> dumping messages with MSILO \n"); >>>> # MSILO - dumping user's offline messages to user >>>> if (m_dump("$fu")) { >>>> xlogl("L_NOTICE", "MSILO: offline messages for $fu dumped - if >>>> they were \n"); >>>> } >>>> else { >>>> xlogl("L_NOTICE", "MSILO: no offline messages dumped \n"); >>>> }; >>>> } >>>> >>>> >>>> >>>> This means dump the message only when it is not re-registration. >>>> >>>> >>>> >>>> Thanks >>>> >>>> >>>> >>>> Regards >>>> >>>> Muhammad Zaka >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> *From:* sr-users [mailto:sr-users-bounces@lists.kamailio.org] *On >>>> Behalf Of *Abdul Basit >>>> *Sent:* 08 April 2018 18:43 >>>> *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; >>>> Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> >>>> *Subject:* Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive >>>> delivery >>>> >>>> >>>> >>>> Dear Team, >>>> >>>> Still stuck with the issue. Kindly suggest where i did wrong. >>>> >>>> >>>> >>>> >>>> -- >>>> regards, >>>> >>>> >>>> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >>>> >>>> >>>> >>>> On 6 April 2018 at 20:50, Abdul Basit <basit.engg(a)gmail.com> wrote: >>>> >>>> Hi team, >>>> >>>> I am facing an issue while setting up IM server based on kamailio. >>>> >>>> SIP-SIP online messaging is fine. For offline messaging i am using >>>> msilo module which is storing offline messages in 'silo' table. >>>> >>>> >>>> >>>> As B-party come online ... it receive offline messages stored in >>>> database. But >>>> >>>> *Problem is*.. these messages will deliver to B-party as many time it >>>> REGISTER online. >>>> >>>> >>>> >>>> How can I configure msilo module to mark messages delivery status? >>>> or how can i delete the delivered messages so that there will be no >>>> chance of delivering them again on next REGISTER? >>>> >>>> Message parts of kamailio.cfg are >>>> .... >>>> /* add local domain aliases */ >>>> alias="192.168.10.18" >>>> port=9060 >>>> >>>> >>>> # ----- msilo params ----- >>>> modparam("msilo", "db_url", DBURL) >>>> modparam("msilo", "db_table", "silo") >>>> modparam("msilo", "sc_mid", "id") >>>> modparam("msilo","from_address","sip:registrar@192.168.10.18:9060") >>>> modparam("msilo","contact_hdr","Contact: registrar@192.168.10.18:9060;m >>>> silo=yes\r\n") >>>> modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") >>>> >>>> ........ >>>> >>>> route[MSILO_MESSAGE] { >>>> if (!is_method("MESSAGE")) return; >>>> xlogl("L_NOTICE", "[$rm] r-uri: $ru d-uri: $du \n"); >>>> if (lookup("location")) { >>>> xlogl("L_NOTICE", "[$rm] -> loaction found. Relying message >>>> [[$rb]] to the UAC \n"); >>>> t_relay(); >>>> } else { >>>> if (!t_newtran()) { >>>> xlogl("L_ERROR", "[$rm] -> ERROR creating new transaction >>>> \n"); >>>> sl_reply_error(); >>>> exit; >>>> }; >>>> xlogl("L_NOTICE", "[$rm] -> Storing MSILO_MESSAGE \n >>>> [$ru]-[$tu]-[$ou] \n "); >>>> if (m_store("$ru")) { >>>> xlogl("L_NOTICE", "MSILO: offline message stored! \n"); >>>> if (!t_reply("202", "Accpeted and Strored")) { >>>> xlogl("L_ERROR", "MSILO: Err sending 202 reply... \n"); >>>> sl_reply_error(); >>>> } >>>> } else { >>>> xlogl("L_ERROR", "MSILO: offline message NOT stored... \n"); >>>> if (!t_reply("503", "Service Unavailable")) { >>>> sl_reply_error(); >>>> } >>>> } >>>> } >>>> exit; >>>> } >>>> >>>> ..... >>>> >>>> # Handle SIP registrations >>>> route[REGISTRAR] { >>>> if (!is_method("REGISTER")) return; // discard all non-REGISTER >>>> methods >>>> >>>> if(isflagset(FLT_NATS)) { >>>> setbflag(FLB_NATB); >>>> #!ifdef WITH_NATSIPPING >>>> setbflag(FLB_NATSIPPING); //# do SIP NAT pinging >>>> #!endif >>>> } >>>> $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto; >>>> xlogl("L_INFO", "[$rm] -> UAC actual location $avp(received) \n"); >>>> if (!save("location",0x04)) { >>>> xlogl("L_ERROR", "[$rm] -> Unable to save user location \n"); >>>> sl_reply_error(); >>>> } >>>> else { >>>> xlogl("L_INFO", "[$rm] -> dumping messages with MSILO \n"); >>>> # MSILO - dumping user's offline messages to user >>>> if (m_dump("$fu")) { >>>> xlogl("L_NOTICE", "MSILO: offline messages for $fu dumped - >>>> if they were \n"); >>>> } >>>> else { >>>> xlogl("L_NOTICE", "MSILO: no offline messages dumped \n"); >>>> }; >>>> } >>>> exit; >>>> } >>>> >>>> Any idea? >>>> >>>> >>>> -- >>>> regards, >>>> >>>> >>>> abdul basit >>>> >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >> >

2 1

Limiting Kamailios accepted methods
by Gholamreza Sabery 12 Apr '18

12 Apr '18

Hi, I want to limit Kamailios accepted methods to REGISTER, INVITE and their related methods like BYE and CANCEL. How to do this in Kamailio? Is there a standard way or I should use is_method function? Regards

3 4

Re: [SR-Users] Is 't_newtran()' needed when using reliable TCP/TLS transport?
by Cody Herzog 11 Apr '18

11 Apr '18

Another thought... Perhaps 'async_task_route()' automatically creates a transaction, which is why I was having problems with the calls to 't_newtran()' and 't_release()' after the route had been changed to execute through 'async_task_route()'. If so, then maybe it's safe for my async presence route to not call 't_newtran()' and 't_release()'. Thanks.

1 0

Re: [SR-Users] Is 't_newtran()' needed when using reliable TCP/TLS transport?
by Cody Herzog 11 Apr '18

11 Apr '18

Thanks for the quick reply, Alex. I will try to provide more context information. The route in question is related to presence. Our usage of t_newtran() is very similar to the following presence examples: https://github.com/kamailio/kamailio/blob/master/test/unit/presence.cfg https://lists.kamailio.org/pipermail/sr-users/2016-April/092544.html https://www.kamailio.org/dokuwiki/doku.php/presence:configuration-file (see route[2]) I believe our config was probably based off one of those. In that context, with reliable transport, is 't_newtran()' useful? Thanks.

1 0

Is 't_newtran()' needed when using reliable TCP/TLS transport?
by Cody Herzog 11 Apr '18

11 Apr '18

Hello. I have read that the primary use of 't_newtran()' is to handle SIP retransmissions. Given that, is there any reason to use it when operating over TCP/TLS? I recently changed a route to operate asynchronously through 'async_task_route()'. When I did, calls to 't_newtran()' started failing. I'm hoping that I can safely remove all the calls to 't_newtran()' and the matching calls to 't_release()' on this async route, because I am using TLS as the transport. Does that sound correct? Thanks.

2 1

Kamailio logs in RHEL 7 environment
by igor.potjevlesch＠gmail.com 11 Apr '18

11 Apr '18

Hello ! Since I moved my OS to RHEL 7, the logs management has changed. I cannot have all logs in /var/log/messages as I used to have. Indeed, there is journald/rsyslog which blocked logs to be written after hundreds/thousands of messages. I tried to set these values to 0: vim /etc/systemd/journald.conf [Journal] [.] RateLimitInterval=0 RateLimitBurst=0 But the problem is still present. Anyone has already solved this problem? Regards, Igor.

2 2

Default AUTH route potential issue?
by Володимир Іванець 11 Apr '18

11 Apr '18

Hello all! I'm using Kamailio 5.1.0 on my testing machine. Configuration includes slightly modified AUTH route from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb route[AUTH] { xlog("L_DBG", "== TRACE. AUTH\n"); # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { xlog("L_DBG", "== TRACE. AUTH. Not relaying. Exiting.\n"); sl_send_reply("403","Not relaying"); exit; } if(isflagset(TRUSTEDIP)) { xlog("== TRACE. AUTH. TRUSTEDIP. Returning.\n"); return; } if (is_method("REGISTER") || from_uri==myself) { xlog("L_DBG", "== TRACE. AUTH. Method REGISTER\n"); # authenticate requests if (!auth_check("$fd", "sipusers", "1")) { auth_challenge("$fd", "0"); xlog("L_DBG", "== TRACE. AUTH. Exiting.\n"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) { xlog("L_DBG", "== TRACE. AUTH. Method is not REGISTER|PUBLISH\n"); consume_credentials(); } } xlog("L_DBG", "== TRACE. AUTH. Returning.\n"); return; } I opened port UDP/5060 to everyone today and started receiving some SIP requests. Most INVITEs were stopped by *auth_challenge* but then I received this one: 2018/04/11 16:32:44.385689 38.91.106.211:5069 -> 172.16.30.205:5060 INVITE sip:100@MY_PUB_IP_ADDRESS SIP/2.0 v: SIP/2.0/UDP 38.91.106.211:5060;branch=z9hG4bK-929181129;rport Content-Length: 0 f: "pbx"<sip:100@1.1.1.1>;tag=3535306165633930313363340131373533363938373235 i: 757925348661465531074812 m: sip:100@38.91.106.211:5069 Accept: application/sdp CSeq: 1 INVITE t: "pbx"<sip:100@1.1.1.1> Max-Forwards: 70 ... and it came through AUTH route. Below are two fragments of Kamailio log: Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. INVITE From: sip:100@1.1.1.1 (IP:38.91.106.211:5069) Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. To: sip:100@1.1.1.1 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: pv [pv_core.c:1286]: pv_get_dsturi(): no destination URI Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. Destination URI : <null> Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. SIP Request header : sip:100@MY_PUB_IP_ADDRESS Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: pv [pv_core.c:966]: pv_get_useragent(): no User-Agent header Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. User Agent header : <null> **************************************************************************************************** Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. request_route ==> AUTH Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. AUTH Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==13 && [1.1.1.1] == [172.16.30.205] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/forward.c:412]: check_self(): host != me Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==13 && [1.1.1.1] == [172.16.30.205] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/forward.c:412]: check_self(): host != me Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 13==9 && [ MY_PUB_IP_ADDRESS ] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 13==13 && [ MY_PUB_IP_ADDRESS ] == [172.16.30.205] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 13==9 && [ MY_PUB_IP_ADDRESS ] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. AUTH. Returning. As you can see all tests failed to catch this INVITE request and Kamailio continued processing it. And I'm now wondering what would be the best way to identify such packet. Thanks.

2 1

[SOLVED]: rtpengine no voice when working between DTLS-SRTP endpoints
by Aqs Younas 11 Apr '18

11 Apr '18

DTLS=passive solved it. Thanks guys. On Tue, 10 Apr 2018, 11:37 pm Aqs Younas, <aqsyounas(a)gmail.com> wrote: > Sometimes, I see below logs in RTP engine. > > > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Received command 'answer' from > 127.0.0.1:44933 > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: answer time = 0.000163 sec > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Replying to 'answer' from > 127.0.0.1:44933 > [1523383161.279950] ERR: [24b95195-3da3-4e12-8400-5fcf908183e5 port > 8268]: SRTP output wanted, but no crypto suite was negotiated > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8268]: SRTP output wanted, but > no crypto suite was negotiated > Apr 10 17:59:25 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8268]: Confirmed peer address > as 72.214.35.171:64834 > [1523383171.481023] ERR: [24b95195-3da3-4e12-8400-5fcf908183e5 port > 8269]: SRTCP output wanted, but no crypto suite was negotiated > Apr 10 17:59:31 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8269]: SRTCP output wanted, but > no crypto suite was negotiated > Apr 10 17:59:31 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8269]: Confirmed peer address > as 72.214.35.171:50108 > [1523383176.025296] ERR: [24b95195-3da3-4e12-8400-5fcf908183e5 port > 8268]: SRTP output wanted, but no crypto suite was negotiated > Apr 10 17:59:36 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8268]: SRTP output wanted, but > no crypto suite was negotiated > [1523383186.000280] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error: 1 (read timeout expired) > [1523383186.000335] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error on local port 8248 > [1523383186.000419] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error: 1 (read timeout expired) > [1523383186.000429] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error on local port 8249 > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error: 1 (read timeout > expired)* > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error on local port 8248* > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error: 1 (read timeout > expired)* > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error on local port 8249* > *Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Closing call due to timeout* > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Final packet stats: > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --- Tag > 'f6d2237d-f960-4542-b138-f39a7fb52770', created 1:32 ago for branch '', in > dialogue with '6bdf30d1-2da6-4b6d-b917-aaa720c9c1fa' > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: ------ Media #1 (audio over > UDP/TLS/RTP/SAVP) using unknown codec > *Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port 209.182.216.71:8288 > <http://209.182.216.71:8288> <> 100.84.103.245:4002 > <http://100.84.103.245:4002> , SSRC 0, 0 p, 0 b, 0 e, 92 ts* > *Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port 209.182.216.71:8289 > <http://209.182.216.71:8289> <> 100.84.103.245:4003 > <http://100.84.103.245:4003> (RTCP), SSRC 0, 0 p, 0 b, 0 e, 92 ts* > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --- Tag > '6bdf30d1-2da6-4b6d-b917-aaa720c9c1fa', created 1:32 ago for branch '', in > dialogue with 'f6d2237d-f960-4542-b138-f39a7fb52770' > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: ------ Media #1 (audio over > UDP/TLS/RTP/SAVP) using G722/8000 > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port > 209.182.216.71:8268 <> 72.214.35.171:64834, SSRC 653128b4, 935 p, > 160820 b, 0 e, 60 ts > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port > 209.182.216.71:8269 <> 72.214.35.171:50108 (RTCP), SSRC 653128b4, 3 p, > 278 b, 0 e, 60 ts > > > Any suggestion what might be happening? > > Br, Aqs. > > On 10 April 2018 at 22:59, Aqs Younas <aqsyounas(a)gmail.com> wrote: > >> I could see SRTP packets coming from one device but they never leave >> rtpeninge. >> >> I put a link to Pastebin containing a call trace with the hope that >> someone might help me out. >> >> I could provide more info if required. >> >> https://pastebin.com/tYVpFQAh >> >> Br, Aqs. >> >> >> On 10 April 2018 at 01:39, Aqs Younas <aqsyounas(a)gmail.com> wrote: >> >>> Greetings list, >>> >>> I am trying to make two endpoints talking on DTLS-SRTP. But I hear on >>> audio. >>> >>> Things work perfectly fine if I use RTP or SRTP with TLS. >>> >>> Endpoints are pjsip based application not webrtc based clients. >>> >>> >>> Below are logs from rtpengine. I hope someone could point out amiss. >>> >>> Apr 9 20:02:43 centos-1024mb-nyc-02 rtpengine[58438]: INFO: >>> [66d2da58-21fe-48bd-9999-a1f3a22afa6d]: --------- Port >>> 209.182.216.71:8176 <> 72.214.35.171:63577, SSRC 1234c6eb, 641 p, >>> 110252 b, 0 e, 60 ts >>> Apr 9 20:02:43 centos-1024mb-nyc-02 rtpengine[58438]: INFO: >>> [66d2da58-21fe-48bd-9999-a1f3a22afa6d]: --------- Port >>> 209.182.216.71:8177 <> 72.214.35.171:63056 (RTCP), SSRC 1234c6eb, 4 >>> p, 372 b, 0 e, 60 ts >>> Apr 9 20:12:24 centos-1024mb-nyc-02 rtpengine[58438]: INFO: Version >>> git-master-3ef300b shutting down >>> Apr 9 20:12:37 centos-1024mb-nyc-02 rtpengine[58958]: INFO: Generating >>> new DTLS certificate >>> Apr 9 20:12:37 centos-1024mb-nyc-02 rtpengine[58959]: INFO: Startup >>> complete, version git-master-3ef300b >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Received command 'offer' from >>> 127.0.0.1:57645 >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: NOTICE: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Creating new call >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: offer time = 0.002612 sec >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Replying to 'offer' from >>> 127.0.0.1:57645 >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Received command 'answer' from >>> 127.0.0.1:42309 >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: answer time = 0.000220 sec >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Replying to 'answer' from >>> 127.0.0.1:42309 >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: DTLS: Peer certificate >>> accepted >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: DTLS-SRTP successfully >>> negotiated >>> Apr 9 20:13:57 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: SRTP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:00 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: Confirmed peer address >>> as 72.214.35.171:58634 >>> Apr 9 20:14:07 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8001]: SRTCP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:07 centos-1024mb-nyc-02 rtpengine[58959]: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08] version=2, padding=0, count=1, >>> payloadtype=200, length=12, ssrc=2045607967, ntp_sec=1379282714, >>> ntp_fractions=439054259, rtp_ts=1838072137, sender_packets=3366262813, >>> sender_bytes=2383498210, ssrc=815258372, fraction_lost=96, >>> packet_loss=13713522, last_seq=3314313929, jitter=2878956247, >>> last_sr=2456273253, delay_since_last_sr=3351655681 >>> Apr 9 20:14:07 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8001]: Confirmed peer address >>> as 72.214.35.171:57732 >>> Apr 9 20:14:12 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: SRTP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:17 centos-1024mb-nyc-02 rtpengine[58959]: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08] version=2, padding=0, count=1, >>> payloadtype=200, length=12, ssrc=2045607967, ntp_sec=2811881700, >>> ntp_fractions=4080266212, rtp_ts=371429680, sender_packets=958830616, >>> sender_bytes=2579186043, ssrc=1909756377, fraction_lost=174, >>> packet_loss=11416637, last_seq=3106722675, jitter=758758394, >>> last_sr=2663618457, delay_since_last_sr=1399181077 >>> Apr 9 20:14:27 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: SRTP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:27 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8001]: SRTCP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:27 centos-1024mb-nyc-02 rtpengine[58959]: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08] version=2, padding=0, cou >>> >>> >>> It is how I have programmed it in my Kamailio configuration. >>> >>> ON INVITE >>> >>> rtpengine_offer("replace-origin replace-session-connection ICE=remove >>> UDP/TLS/RTP/SAVP"); >>> >>> ON 200-ok >>> >>> rtpengine_answer("replace-origin replace-session-connection ICE=remove >>> UDP/TLS/RTP/SAVP"); >>> >>> >>> Best Regards, >>> >>> Aqs Younas >>> >> >> >

1 0

Is it possible to alter SDP content before forwarding the packet?
by Wilkins, Steve 11 Apr '18

11 Apr '18

Hello all, I was curious whether or not it possible to alter SDP content before forwarding the packet? Thank you all as always!

2 1

Re: [SR-Users] Reply Destination
by Duarte Rocha 11 Apr '18

11 Apr '18

It works. Thanks a lot!!

1 0

Reply Destination
by Duarte Rocha 11 Apr '18

11 Apr '18

Hey Is there any variable that retrieves me the destination of a reply? I've used the To header for the domain, but i can't seem to get the destination port. Any help ? Thanks

2 1

Need installation steps for installing kamailio 5.1.2 with debian packages
by vinay kumar 11 Apr '18

11 Apr '18

Hi All, can anyone share me the installtion steps for kamailio 5.1.2 with debian packages on ubuntu. i have used kamailio 4.3.6 with polaris configuration. I want to upgrade latest version of kamailio. And i have installed latest debian packages and i have used polaris scripts to bringup kamailio and its failing with status kamailio service with Result resources. if i use latest debian packages , are polaris configurations are compatible with latest packages?? Please help me out. Thanks in advance. Regards, vinay

1 0

rtpengine
by volga629＠networklab.ca 11 Apr '18

11 Apr '18

Hello Everyone, What flags available on latest rtpegnine to acomplish transcoding and codec manipulation ? Slava.

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