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37537 discussions

rtpengine no voice when working between DTLS-SRTP endpoints
by Aqs Younas 10 Apr '18

10 Apr '18

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 … [View More]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 [View Less]

1 2

RTPengine - Transcoding example available?
by Carsten Bock 10 Apr '18

10 Apr '18

Hi, Quick (and simple) question: Can someone provide a working RTPengine request, to enable Transcoding? I know, this is possible, but I thought before looking into deep, how this should be configured for the rtpengine_offer() command, could someone provide an example? Thanks, Carsten

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Re: [SR-Users] First Invite
by Duarte Rocha 10 Apr '18

10 Apr '18

Thank you very much!! That seems to be the solution. Cheers

1 0

First Invite
by Duarte Rocha 10 Apr '18

10 Apr '18

Greetings, How can i identify the first Invite received by Kamailio in a dialog. I'm checking for the To-Tag in order to identify the first one. However, if the UA retries the same INVITE (because of a timeout for example), it won't have To Tag and it won't be the first INVITE that i'm receiving in that dialog. Is there any flag or variable that i can use to identify the first Invite received? Thanks in advance

2 1

Kamailio ds_ping_interval Performance
by Atul Thosar 10 Apr '18

10 Apr '18

Hi All, I am using Kamailio ** *v4.4.x* to load balanced traffic to FreeSWITCH servers. I have query regarding ds_ping_interval and ds_probing_threshold. We have very high traffic (around 200-400 (CPS) calls per sec) hitting on Kamailio which then distribute it to 2-3 FreeSWITCH servers. What is the optimal value should I set to ds_ping_interval and ds_probing_threshold? If I set ds_ping_interval=2 and ds_probing_threshold=1 then in every 2 sec, I would come to know if my … [View More]

6 8

Issues on fresh install
by Worth Morrison 10 Apr '18

10 Apr '18

what does it mean when my siremis login looks like this after an install that seems to have been successful? http://pasteall.org/pic/show.php?id=2b72124d5046e2250ad41d14c722c54d the page is completely non-functional. Thanks, *Worth Morrison*

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Calling a mobile SIP client, which needs to update the registration
by Ulrich Henning 09 Apr '18

09 Apr '18

Hi everybody, I am trying to call a sip client on a mobile cellphone, which is registered fine at my Kamailio instance. Everything is working fine if the phone is awake and the mobile app is not sleeping (e.g. energy saving by OS). If the device is sleeping, the cellphone gets a wakeup request call via apple push kit and wakes the app. Currently my app is doing a Register after each wakeup, because the device does not know if the cellular network changed (external IPv4 address with Carrier … [View More]

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MSILO: SIP stored offline MESSAGE repetitive delivery
by Abdul Basit 09 Apr '18

09 Apr '18

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 … [View More]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; msilo=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 [View Less]

4 6

Problem capturing packets from kamailio to homer sip-capture
by Arik Halperin 07 Apr '18

07 Apr '18

Hello, My kamailio is listening to TCP, so when I try to duplicate packets to homer, It tries to send them as TCP. From the log I see: Apr 5 08:21:45 kamprod /usr/local/sbin/kamailio[16309]: ERROR: <core> [core/forward.c:181]: get_out_socket(): no socket found Apr 5 08:21:45 kamprod /usr/local/sbin/kamailio[16309]: ERROR: <core> [core/forward.c:183]: get_out_socket(): no corresponding socket found for(udp:10.168.10.229:5090) Apr 5 08:21:45 kamprod /usr/local/sbin/kamailio[16309]… [View More]

4 3

Dialog Profile Value Name Limit
by Duarte Rocha 06 Apr '18

06 Apr '18

Greetings, I'm using the Dialog module and the profile functions in order to limit the number of channels that a Calling Number can use. I'm using profile with values and the number in E164. However it doesn't work correctly and the value are not correctly updated. I've tried to give a a value with only 5 digits and everything seems to work fine. Does the set_dlg_profile and unset_dlg_profile have a size limit for it's value ? Thanks

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