sr-users December 2016

sr-users@lists.kamailio.org

71 participants
101 discussions

rtpengine destroys voice branch on multiple branch call
by Yuriy Gorlichenko 27 Dec '16

27 Dec '16

Hi. I have 2 endpoints 1001 and 1002 1001 registered from 1 device 1002 registered from 2 devices at the same time When I called to 1002 kamailio makes 2 branches rtpengine_manage command called from branch rout for handling every branch directly (it can be different endoint types (ws/tls/udp) for each branch) When i picking up at the 1002 on one device server sends CANCEL (answered elswhere) to another devices of 1002 rtpengine_manage reacts at the CANCEL and deletes this brach after 30 seconds My question is: What is solution for use rtpengine_manage and call branches together?

1 0

set_body_multipart and append_body_part
by Diego Nadares 26 Dec '16

26 Dec '16

Hi guys, I'm trying to append a content type to a message that has only one content type. I tried using just set_body_multipart and now I'm using set_body multipart and append_body_part with no success too. The result is a message with multipart but has "two" sdp and no gtd. It has the old sdp with no ip changed for rtpproxy and the new sdp with ip changed BUT with no content type. And the message has not the end boundary (--unique-boundary-1.) Kamailio log has an info and a warning Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: INFO: <core> [msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params <application/sdp> Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: WARNING: <core> [msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries error kamailio.cfg ...... route[RELAY]{ $avp(ch) = 1 .... if ($avp(ch)){ set_body_multipart(); msg_apply_changes(); $var(gtd) ="IAM,\r\nGCI,asdafsdfasd\r\n\r\n"; append_body_part("$var(gtd)", "application/gtd", "signal;handling=optional"); } .... } Ngrep U 2016/12/26 11:36:22.496634 172.16.213.21:5060 -> 172.16.213.38:5060 ... Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. U 2016/12/26 11:36:22.688458 172.16.213.38:5060 -> 172.16.208.11:5060 .... Content-Type: multipart/mixed;boundary="unique-boundary-1". Mime-Version: 1.0. Remote-Party-ID: <sip:1152565064@172.16.213.38:5060 >;party=calling;privacy=off;screen=no. . --unique-boundary-1. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. . --unique-boundary-1. v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 58724 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=sendrecv. a=rtcp:58725. a=ice-ufrag:hhXDwrco. a=ice-pwd:5temIXP8veOGuD8Dsllm5HXkWU. Any help will be very very appreciated! Thanks in advance. Diego

2 4

help configure dispatcher in Amazon ec2
by huan nguyen duy 26 Dec '16

26 Dec '16

Hi, I’m very appreciated if someone can figure out this problem for me. My aim is to setup HA Kamailio using Dispatcher module in Amazon EC2, system like this: soft phone 1(Linphone) —————> | Load balancer —————> | Sip Server huancomputer | private IP: 172.31.11.35 | 172.31.13.113 soft phone 2(Linphone) —————> | public IP: 35.166.179.244 | 35.166.152.104 huannguyen | | The dispatcher.lst config file in /usr/local/etc/kamailio as below: # gatewayss 1 sip:172.31.13.113:5060;transport=udp <sip:172.31.13.113:5060;transport=udp> 0 0 abc=xyz User records subscriber table: ——————————————————————————————————————————————————— id | username | domain ——————————————————————————————————————————————————— 70 | nguyenhuan | nguyenhuan(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> 71 | huancomputer | huancomputer(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> Location records from location table: ——————————————————————————————————————————————————— username | domain | contact ——————————————————————————————————————————————————— huancomputer | NULL | sip: huancomputer(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:64166;transport=udp nguyenhuan | NULL | sip: nguyenhuan(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:33267;transport=udp But soft phone 1 cannot make a call to soft phone 2, below is data is extracted from wireshark: Load balancer: -------------------------------- No. Time Source Destination Protocol Length Info 177 4.463129 14.161.41.70 172.31.11.35 SIP/SDP 1433 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 179 4.463367 172.31.11.35 14.161.41.70 SIP 452 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 180 4.463421 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1215 32.552942 14.161.41.70 172.31.11.35 SIP 1105 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1216 32.553170 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1219 32.555588 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1220 32.555706 172.31.11.35 14.161.41.70 SIP 593 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1609 42.603150 14.161.41.70 172.31.11.35 SIP 1113 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1610 42.603364 172.31.11.35 172.31.13.113 SIP 1269 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1613 42.605800 172.31.13.113 172.31.11.35 SIP 749 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1614 42.605932 172.31.11.35 14.161.41.70 SIP 599 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1688 44.425237 172.31.11.35 14.161.41.70 SIP 471 Status: 408 Request Timeout | No. Time Source Destination Protocol Length Info 1689 44.425285 172.31.11.35 172.31.13.113 SIP 491 Request: CANCEL sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1691 44.426018 172.31.13.113 172.31.11.35 SIP 519 Status: 200 canceling | Data from Sip Server: ------------------------------- No. Time Source Destination Protocol Length Info 237 6.242928 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 243 6.244409 172.31.13.113 172.31.11.35 SIP 602 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 244 6.244541 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 288 6.710414 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 328 7.710454 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 396 9.710411 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 547 13.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 688 17.710406 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 835 21.710381 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 985 25.710385 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1130 29.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1279 33.710400 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1303 34.332395 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1312 34.334415 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1459 37.710402 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1609 41.710395 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | Huan Nguyen, --------------------------------------- Mobile app developer Phan Thi Company Sai Gon, VietNam Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 1

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 24 Dec '16

24 Dec '16

While waiting for Santa, it's a good time to also express my thanks and greetings to all the friends, developers and community members that made 2016 another amazing year for Kamailio project. Enjoy the winter holidays together with your beloved ones! Merry Christmas! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Merry Christmas to all Kamailians
by Olle E. Johansson 24 Dec '16

24 Dec '16

Friends, Trying to calm down and prepare for Christmas with the family. It’s been another great year for Kamailio and I’m proud to be part of the Kamailio development community. We’ve made great releases, had a great conference and overall done good stuff :-) For all of you that celebrate Christmas - a Merry Christmas! To the rest of you: Happy Holidays! A big Thank You to all developers and a special giga-Thank You to Daniel for all the time you spend working with the code, helping out in discussions on the lists and in the bug tracker. And to all of you in the community - Thank You for being part of this! Greetings from a cold Sweden, currently without snow in my area just north of Stockholm. We celebrate on Christmas Eve (tomorrow) so there’s a lot of food being prepared right now and the last minute shopping is just about to start... Keep your Kamailio running! /Olle PS. If you are not currently following the project on Twitter - it’s time! Find us as @kamailioproject

3 2

baresip version 0.5.0 released
by Alfred E. Heggestad 23 Dec '16

23 Dec '16

Hi, my apologies for the off-topic posting but I think this could be interesting for the SIP Community. Baresip version 0.5.0 is now released, you can find the announcement email here: http://lists.creytiv.com/pipermail/re-devel/2016-December/001154.html Merry Xmas from Norway! /alfred

1 0

Re: [SR-Users] rtjson + mhomed and record route (corrected)
by Diego Nadares 23 Dec '16

23 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The call is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header? ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } .... Thanks in advance. Diego 2016-12-22 12:08 GMT-03:00 Diego Nadares <dnadares(a)gmail.com>: > Hi list, > > We have two eth, one with a private ip and the other with a public ip. We > have MHOMED configured. The call comes from private network and then is > routed to a public network via rtjson routing. The is routed but the thing > is that the record route is being set with the private one (It's the same > when calls come from public and is routed to a private network, record > route has the public and not the private). > > With mhomed configured shouldn't Record Route be updated as VIA header > > ".... When activated, sip-router will select a socket that can reach the > destination (to be able to connect to the remote address). (sip-router > opens a UDP socket to the destination, then it retrieves the local IP which > was assigned by the operating system to the new UDP socket. Then this > socket will be closed and the retrieved IP address will be used as IP > address in the Via/Record-Route headers)..." > > This is part of ngrep > > U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 > INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. > To: <sip:1111111111@172.16.213.38:5060>. > Call-ID: 1-13024(a)172.16.213.21. > CSeq: 1 INVITE. > Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. > .... > > U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > .... > > U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 > INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. > *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* > Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f > Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 > > This is my cfg > > mhomed=1 > > request_route { > > > # per request initial checks > route(REQINIT); > > # NAT detection > #route(NATDETECT); > > # CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > route(RELAY); > } > exit; > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # handle retransmissions > if(t_precheck_trans()) { > t_check_trans(); > exit; > } > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are > routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) { > record_route(); > } > > > Thanks in advance. > > Diego. > >

1 1

kamailio dispatcher call flow example
by huan nguyen duy 23 Dec '16

23 Dec '16

Hi, I want to load balance multiple kamailio server using dispatcher module but not success at configuration. I’m not clearly how it work. Someone can show me an example call flow like below i’m very appreciated. Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| | | | 407 F2 | | | |<---------------| | | | ACK F3 | | | |--------------->| | | | INVITE F4 | | | |--------------->| INVITE F5 | | | 100 F6 |--------------->| INVITE F7 | |<---------------| 100 F8 |--------------->| | |<---------------| | | | | 180 F9 | | | 180 F10 |<---------------| | 180 F11 |<---------------| | |<---------------| | 200 F12 | | | 200 F13 |<---------------| | 200 F14 |<---------------| | |<---------------| | | | ACK F15 | | | |--------------->| ACK F16 | | | |--------------->| ACK F17 | | | |--------------->| | Both Way RTP Media | |<================================================>| | | | BYE F18 | | | BYE F19 |<---------------| | BYE F20 |<---------------| | |<---------------| | | | 200 F21 | | | |--------------->| 200 F22 | | | |--------------->| 200 F23 | | | |--------------->| | | | | Huan Nguyen, ——————————————————— Mobile app developer Phan Thi Company Sai Gon, VietNam

2 1

rtpproxy/rtpengine. Help to understand.
by Igor Olhovskiy 22 Dec '16

22 Dec '16

Hi! Issue I can’t figure out. Or all working ok and that’s just me who not understands. I have situation softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX (111.222.3.4) All addresses are on same public network. rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3 I want all media goes through rtpproxy. So, in INVITE and ACK I call rtpproxy_manage(). But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 in SDP, but not 10.0.0.3 If I use scheme, where client in some other NATted network (like 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working) So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or I’m missing something? If any - can provide pcap’s with both cases. Regards, Igor

3 5

rtjson + mhomed and record route
by Diego Nadares 22 Dec '16

22 Dec '16

1 0

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sr-users December 2016