sr-users February 2018

sr-users@lists.kamailio.org

90 participants
110 discussions

RTPEngine Transcoding
by Social Boh 14 Feb '18

14 Feb '18

Hi List, I just read on the RTPEngine page the Proxy RTP now support transcoding: https://github.com/sipwise/rtpengine#transcoding is there a plan to support it? Regards -- --- I'm SoCIaL, MayBe

5 10

RTPEngine QoS statistics and leg labelling
by Alex Balashov 14 Feb '18

14 Feb '18

Hello, I am excited about the new media statistics from RTPEngine, exposed conveniently into PVs, e.g. jitter, packet loss, MOS, time offset of lowest and highest MOS scores, etc. Great work! I am, however, having a bit of trouble understanding how to properly apply this new functionality, as it is not clearly documented (IMHO): 1. What is the meaning of "global" statistics which do not take into account an "A" and a "B" side? That is to say, if I have a packet loss or a jitter or MOS score statistic using just the generic toplevel values, e.g. https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… What does that mean? A functional call ordinarily consists of an inbound and outbound RTP stream direction relative to the endpoints on both sides of RTPEngine. Inbound A --> server --> Outbound A Outbound B <-- server <--- Inbound B Where do the "global" stats situate themselves in relation to all this? 2. I am given to understand that the statistics for the "A" and "B" sides of a bidirectional RTP stream can be separated based on an arbitrary label, using these PVs: https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… How do I use it properly? My intuition was to try something like this: rtpengine_offer("replace-origin replace-session-connection ICE=remove label=outbound"); And in the answer/manage handler on the SDP answer: rtpengine_answer("replace-origin replace-session-connection ICE=remove label=inbound"); Later, when processing a BYE and obtaining subsequent media statistics, I tried this: if(check_route_param("proxy_media=yes")) { $avp(rtpengine_a_label) = 'outbound'; $avp(rtpengine_b_label) = 'inbound'; rtpengine_delete(); route(LOG_RTPENGINE_STATS); } But this only gets me stats for the "A" leg in the JSON object I emit: { "server_addr": "udp:127.0.0.1:5050", "a_label": "outbound", "b_label": "inbound", "a_leg": { "min": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "max": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "avg": { "mos": "4.4", "packetloss": "0", "jitter": "0", "roundtrip": 0, "samples": 2 } }, "b_leg": { "min": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "max": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "avg": { "mos": "<null>", "packetloss": "<null>", "jitter": "<null>" } } } I also tried rtpengine_delete("label=inbound") (and outbound) individually. Some clarity on how this all fits together would be appreciated. :-) Many thanks to Sipwise for great work on it! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 14

How to add a new module to existing Kamailio Server?
by Arun NV 14 Feb '18

14 Feb '18

I am using a *Kamailio 4.4* server. And I would like to install *http_async_client* module to this server. I have looked in the module directory(*/usr/local/lib64/kamailio/modules*) to check whether this *http_async _client* module already exists there, but it was not there. And I searched for a solution and only found <http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-modular-install> the method to include new module at the compile time. I could find the modules source code on GitHub <https://github.com/sipwise/kamailio/tree/master/modules/http_async_client>. But I don't know the steps to install a new module to the existing Kamailio Thank you, Arunbalan

2 1

Use RTPProxy in 302 scenario
by Fabian Borot 13 Feb '18

13 Feb '18

I have this scenario Customer --> Kamailio --> Provider1 The route where l route the call to Provider1 is called route["INCOMING"] and I "arm" the replies and failures to : t_on_reply("INCOMING") and t_on_failure("INCOMING"). Provider1 always replies with a 302 Redirect and in the failure_route["INCOMING'] I have this code: if (t_check_status("302")) { get_redirects("*"); t_on_branch("REDIRECT_BRANCH"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL_FORKING"); t_relay(); exit; } In the t_on_failure("SERIAL_FORKING") block I do t_relay() as long as there are more contacts to be processed. Lets say I need to try Provider_2 and Provider_3 failure_route["SERIAL_FORKING"] { xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Reply [$rs]\n"); if (!t_next_contacts()) { exit; } xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Trying next Contact.\n"); t_on_failure("SERIAL_FORKING"); t_relay(); } I want to use RTPProxy in this scenario and I am using the function rtpproxy_manage() to let kamailio handle it automatically. RTPProxy ↑ | | Customer --> Kamailio --> Provider1 | | ↓ Provider N After the 302 response is processed I would like to have the signaling flow "Customer -> Kamailio --> ProviderN" and the RTP flow "Customer --> RTPProxy --> ProviderN" I have tried inserting the function in several places but can't make it work and what I mean is that sometimes the rtpproxy creates the 1st session and then deletes it as soon as I call it the 2nd time with a message similar to this one (log of rtpproxy) : "INFO:416f20b74a58fb75168175e61b2edece@192.168.1.61:5060:rtpp_cmd_delete_ematch: forcefully deleting session 1 on ports 63502/0". Sometimes the kamailio log shows this message: "ERROR: rtpproxy [rtpproxy.c:2536]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" I think I must call rtpproxy_manage() before I send the INVITE to each of the Providers in the Contact list so they receive a SDP offer with an IP:Port combination provided by the RTPProxy (instead of the ones coming from the Customer). That would be in my route["INCOMING'] Then once Provider_N replies with a 183 with SDP or a 200 OK I would call it again to get a new session to be passed on the reply to my customer. But is the part I am struggling with. Should I add a new reply route inside the failure_route["INCOMING'] only if I am dealing with 302 and do it there ? I am out of ideas guys and any idea is welcome

2 2

Can rtpengine change network interface of estabished call?
by Yufei Tao 13 Feb '18

13 Feb '18

Hi, Is it possible to define two pairs of network interfaces for rtpengine, e.g. int/ext and up/down? If so during a call established between interfaces 'int' and 'ext', is it possible to send a re-INVITE from the calling party on interface 'ext', to update the RTP destination to an IP on a different interface 'up', for example? Or does the updated RTP destination have to be on the same interface as when the call was established? I have some success updating the RTP destination if it is on the same interface (different IP:port) however haven't managed to divert it to a different interface. I wonder if it is possible at all or me missing something? Cheers, Yufei

1 0

(no subject)
by Tihomir Culjaga 13 Feb '18

13 Feb '18

Hi, Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=) i have a setup like this: (Kamailio version 5.1) UA > Kamailio > Freeswitch Kamailio forwards registration to freeswitch without saving location. I have configured sip over websockets and im able to register to Freeswitcha via kamailio. Also, i use add_path_received() so kamailio adds a path header to register messages recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809: ------------------------------------------------------------------------ REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 Max-Forwards: 69 To: <sip:1001@192.168.50.60> From: <sip:1001@192.168.50.60>;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90", uri="sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef", qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001 Contact: <sip:1001@192.0.2.110 ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6> ------------------------------------------------------------------------ send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 From: <sip:1001@192.168.50.60>;tag=vq30modpgd To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600 Date: Fri, 09 Feb 2018 21:38:07 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Content-Length: 0 ------------------------------------------------------------------------ Once, registered, im trying to originate a call to the registered endpoint from freeswitch this results in an invite like this: send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549: ------------------------------------------------------------------------ INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0 Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Max-Forwards: 70 From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK To: <sip:1001@192.0.2.110;transport=ws> Call-ID: 8212b700-8885-1236-08b3-000c293ae94e CSeq: 118770326 INVITE Contact: <sip:mod_sofia@192.168.50.65:15060> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 891 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:1002@192.168.50.65 >;party=calling;screen=yes;privacy=off Freeswitch appends a route header to the invite specifying kamailio as route and a received with original uri. well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio my RELAY route is like this route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_ALERT", "Its a Branch, go manage that !!! <$ru>\n"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_ALERT", "its onreply_route, go manage !!! <$ru>\n"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_ALERT", "Its failure_route, go manage !!! <$ru>\n"); t_on_failure("MANAGE_FAILURE"); } } $fs = "udp:192.168.50.60:5060"; xlog("L_ALERT", "Trying to rely !!! <$ru>\n"); if (!t_relay()) { xlog("L_ALERT", "failed to rely !!! <$ru>\n"); sl_reply_error(); } exit; } here is a part of the log: Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:192.168.50.60:5060, to ws:192.0.2.110:5060) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core> [core/msg_translator.c:424]: clen_builder(): content-length: 916 (916) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws> is that because i set a $fu before t_relay() ? any idea how to approach the issue ?

2 4

onend_route() ?
by Wilkins, Steve 13 Feb '18

13 Feb '18

I meant to reply to Kamailio mailing list. From: Wilkins, Steve Sent: Friday, February 9, 2018 12:40 PM To: 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: [SR-Users] t_relay()? Hi Daniel, I am not sure what I am doing wrong but I cannot get the forwarded message to print. I am using onend_route() and trying to print the results using xdbg. I have searched for examples and have had no luck. Thank you From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Friday, February 9, 2018 7:48 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Wilkins, Steve <swwilkins(a)mitre.org<mailto:swwilkins@mitre.org>> Subject: Re: [SR-Users] t_relay()? Hello, before t_relay() you can't see what is going to the network, because some headers are added when knowing target address and local socket (e.g, Via, Record-Route). You can use onsend_route for seeing the request just before writing to the network. https://www.kamailio.org/wiki/cookbooks/devel/core#onsend_route $snd(buf) will give outgoing buffer -- for more see $sndfrom(...) and $sndto(...) vars. Cheers, Daniel On 09.02.18 11:24, Wilkins, Steve wrote: Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com<http://www.asipto.com> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com>

2 1

Request Timeouts
by Wilkins, Steve 13 Feb '18

13 Feb '18

Hello, This is actually a follow up to my last question about some providers calls being dropped. I noticed that when calls are dropped, I see a Kamailio log message "Request Timeout" about 5-10 seconds before the call is dropped. I am wondering what could cause this while a call is in progress. Does Kamailio need to respond to Asterisk with a '200 OK' for an INFO request sent to Kamailio from Asterisk? I am not sure what else could cause this call dropping behavior. Note: It does not appear to be caused from a missing 'ACK'. Thank you, -Steve

3 6

What is the mechanism used for sending push notification from Kamailio, For both IOS and Android?
by Arun NV 13 Feb '18

13 Feb '18

I am creating custom phone client, which uses *Kamailio *as a proxy and it's required to send *push notification* from *Kamailio *to the clients when a call towards the client reached on *Kamailio*. I found a module for APNS here* https://github.com/tvntsr/push <https://github.com/tvntsr/push>*. But I am developing both Android and IOS clients. So What mechanism I can use for the Android side? Is there any module available for *GCM*(Google Cloud Messaging) in Kamailio? Thank you, Arunbalan

2 1

is_audio_on_hold() doesn't support rfc3264
by Julia Boudniatsky 13 Feb '18

13 Feb '18

FYI, After some tests found, that function "*is_audio_on_hold()*" (textops) *supports only rfc 254*3, where "hold" is indicated by setting the "c" destination addresses for the media streams to zero (0.0.0.0). The newest *rfc3264*, where "hold" is indicated by a=sendonly/inactive, *is not supported*. Kamailio version 4.4.6. Best regards, Julia

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sr-users February 2018