sr-users October 2014

sr-users@lists.kamailio.org

101 participants
186 discussions

cSeq increasing
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

Does it possible increase cSeq manually (for example remove and then append headers?) for UAC module when send INVITE messages with Auth, or kamailio have pseudovar for this header?

4 8

UAC INVITE to Provider
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

Hello. I use kamailio for calling to porvider. My providr seccefuully registered from UAC module, but when I try to call through it? it back 401 Unauthorised. I send second try with Digest Auth header at INVITE and it receive me 401 too... I register this provider from asterisk and call succesfully Ok. So i get dump from asterisk This is successfull INVITE: INVITE sip:89126975590@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 17.4.28.7:50600;branch=z9hG4bK5f118681;rport Max-Forwards: 70 From: <sip:gw2@17.4.28.7:50600>;tag=as33192a38 To: <sip:89126975590@sip.provider.com> Contact: <sip:gw2@17.4.28.7:50600> Call-ID: 021088c360a8dbf023bf35560a9daf1e@17.4.28.7:50600 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.6.1 Authorization: Digest username="gw2", realm="provider.com", algorithm=MD5, uri="sip:89126975590@sip.provider.com", nonce="014d80ca", response="67bad8a0c97afc2b6747b471a56bca9f" Date: Wed, 29 Oct 2014 18:50:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 253 v=0 o=root 1098729670 1098729671 IN IP4 17.4.28.7 s=Asterisk PBX 12.6.1 c=IN IP4 17.4.28.7 t=0 0 m=audio 10088 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv Then I get dump from my kamailio (unsuccessfull INVITE) INVITE sip:89126975590@sip.provider.com SIP/2.0 Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as4684d4b9;lr=on> Via: SIP/2.0/UDP sip.myservice.com:5068 ;branch=z9hG4bK600b.1d5ff0fd59d4f3d2a1a06d722c0daa92.2 Via: SIP/2.0/UDP my.aterisk:50600;branch=z9hG4bK2b8d9b09;rport=50600 Max-Forwards: 70 From: <sip:gw2@sip.myservice.com:5068>;tag=as4684d4b9 To: <sip:89126975590@sip.provider.com > Contact:<sip:Vebinar-gw2@sip.myservice.com:5068> Call-ID: 445a7b884aeeab125d91886210c9beb7@sip.myservice.com:50600 CSeq: 102 INVITE User-Agent: SoftSwitch Date: Wed, 29 Oct 2014 22:32:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 Authorization: Digest username="gw2", realm="provider.com", nonce="10129bde", uri="sip:89126975590@sip.provider.com ", response="6d3411b24cbb57ad72271790ec01b453", algorithm=MD5 v=0 o=root 468654998 468654998 IN IP4 1.2.3.4 s=SoftSwitch c=IN IP4 1.2.3.4 t=0 0 m=audio 30104 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30105 I see difference between packetts only at SDP (not inportant things) and at VIA and request route Headers. All other fields identical. So -why Asterisk call successull and Kamailio kall unsuccessfull? What the differense?

2 2

websocket and SIP
by Ricardo Martinez 30 Oct '14

30 Oct '14

Hello. I’m having some problems using websocket to communicate a webRTC client with the SIP world. I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem? Hope that someone could help me. Best Regards, Ricardo Martinez.-

3 5

Verified version of RTPproxy and Kamailio
by Juan Perez 30 Oct '14

30 Oct '14

Can somebody please tell me what would be a verified combination of kamailio and rtpproxy?I mean, for example, kamailio 4.1 and rtpproxy 1.x that can be downloaded from:http://<real_url> I found at least 3 different places where one can download rtpproxy: http://sourceforge.net/p/sippy/rtpproxy/ci/master/tree/ https://github.com/miconda/rtpproxy https://github.com/sippy/rtpproxy I downloaded the one from https://github.com/sippy/rtpproxy and have tested against kamailio 4.1, 4.2 and they are not compatible.It seems that the logi inside the function "send_rtpp_command" is not compatible with the rtpproxy version from https://github.com/sippy/rtpproxy and that causes kamailio to disable rtpproxy and it becomes useless. Second questions: do the feature added to the rtpproxy (parameter -A to use external public IP for devices behind NAT) is compatible with OPENSIPS only? I am about to test with kamailio 3.3, I will update this post later onthank you in advancejp

1 0

kamailio server as Load balancing and Failover
by Nandini madhu 30 Oct '14

30 Oct '14

Dear All, Greetings, I want to implement kamailio server as failover and load balancing. Is there any configuration needed in kamailio.cfg. As I have searched in blogs, For load balancing, Virtual IP is added and also heartbeat software is needed. Please let me Know how to configure kamailio server for load balancing and failover Waiting for response Thank you, Regards, Nandini

3 2

Wrong port in via, from and contact headers
by Eric Koome 30 Oct '14

30 Oct '14

This is my setup: UAC -> firewall (1:1 nat) -> Kamailio:5060 - > asterisk:5080 . | . sip providers I'm running kamailio 4.1.x & ast 11 based on aspitos kamailio configuration tutorial. My problem is port number in via, contact and from headers is 5080 instead of 5060 for calls to provider. Public ip is fine. I have used advertised_port=5060 but no change. Also tried record_route_preset(ip:5060), but no change. I can make calls but I would imagine I would miss acks & and reinvites. Any thoughts on what I'm missing? Eric

1 0

Problem initiating a call with dlg_bridge
by Paul Smith 29 Oct '14

29 Oct '14

I seem to be going round in circles… I am trying to use dlg_bridge() from the dialog module to establish a call between two SIP endpoints. I have tested with Snom phones and linphone soft phone with the same result. I get an outbound call to the first (from) end point, I answer the phone… and then nothing obvious happens. I can get this same result when using kamcmd or from within the kamailio.cfg. For example using kamcmd: kamcmd dlg.bridge_dlg sip:105@mykamailioip sip:106@mykamailioip sip:mykamailioip:5060 I expected this to send an invite from "controller" via my KamailioIP to my registered local subscriber 105 (from), and then send a re-invite to 105 so that 105 creates a call leg to 106 (to). The kamailio.cfg is pretty much the default. I am running 4.2 and rtpengine to proxy the RTP streams. Not sure if it is necessary or useful but I call setflag() at the start of the request_route() to set the Dialog flag. What else should I have included in the kamailio.cfg to make this work? Also is there any way to control the SDP in the initial invite from dlg_bridge. By default I see RTP/AVP, with alaw and ulaw…. I’d like to offer RTP/SAVP swell or instead of RTP/AVP. Thanks

3 7

Trouble with max-forwards
by Mike Dunton 29 Oct '14

29 Oct '14

Readers, I am having issues with passing the max-forward header to my freeswitch service. Here are the sip invites Here is a call coming from a kazoo box though my carrier Into kamailio. Kamailio shows it has Max-Forward after the sanitize check, but when it reaches freeswitch no Max-Forward Header. *KAMAILIO* kamailio[2687]: ERROR: <script>: After Sanitize - [INVITE sip:+1850764####@aio07-bandwidth.tresta-aio.com:5060 <http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0 Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.###:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.59;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.85 <sip%3A%2B18507646071(a)67.231.8.85>> Call-ID: 537168748_70701748(a)67.231.9.# <537168748_70701748(a)67.231.9.59>## CSeq: 1289511105 INVITE *Max-Forwards: 49* Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer X-FS-Support: update_display,send_info Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 *FREESWITCH* recv 1479 bytes from udp/[10.1.13.123]:5060 at 18:31:26.312854: ------------------------------------------------------------------------ INVITE sip:+1850764###@aio07-bandwidth.tresta-aio.com:5060 <http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0 Record-Route: <sip:10.1.13.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 10.1.13.123;branch=z9hG4bKc892.1b308e66.0 Via: SIP/2.0/UDP 67.231.8.195;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.85;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.59:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.# <sip%3A%2B1NUMBERSCRUBED(a)67.231.9.59>##;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.# <sip%3A%2B18507646071(a)67.231.8.85>##> Call-ID: 537168748_70701748(a)67.231.9.# <537168748_70701748(a)67.231.9.59>## CSeq: 1289511105 INVITE MaxForwards would go here? Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no X-AUTH-IP: 67.231.8.195 v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 Here is my kamailio default.cfg http://pastebin.com/vxdFe8n0 Can anyone point me in the right direction, and tell me why freeswitch isn't being passed this header? Both kamailio and freeswitch are on the same box in this example. Thanks for your help. Mike

2 1

404 Not here as response to BYE requests
by Marko Seidenglanz 29 Oct '14

29 Oct '14

Hello, We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC Proxy doing the encryption and NAT Traversal. Everything works as expected, except that BYE Requests sent by the WebRTC Client are not forwarded by Kamailio to Asterisk. We use record routing. Instead Kamailio responds with a "404 Not here". INVITE Headers: Kamailio --> Browser: sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 Via: SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db Max-Forwards: 70 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as32dd9e24 To: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com> Contact: <sip:anonymous@10.240.177.13:5060> Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Wed, 29 Oct 2014 10:04:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676 BYE Headers Browser --> Kamailio: BYE sip:anonymous@anonymous.invalid SIP/2.0 Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 , SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com >;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3 To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as32dd9e24 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 0 BYE Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Reason: Q.850;cause=16 BYE Response Kamailio --> Browser: SIP/2.0 404 Not here Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37 From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com >;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8 To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as0b12a022 Call-ID: 683def08655393622c55299c1a7c92d5@10.240.177.13:5060 CSeq: 0 BYE Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 In the logs I can see the following messages: [loose.c:113]: find_first_route(): No Route headers found [loose.c:929]: loose_route(): There is no Route HF Does anybody know, why Kamailio may respond with 404 Not here? Do I have to send the BYE request directly to Asterisk? Kind regards, Marko

2 3

Kamailio and RTPProxy
by Juan Perez 29 Oct '14

29 Oct '14

I installed Kamailio 4.2 and the latest rtpproxy. When starting Kamailio I see these lines: Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [sr_module.c:590]: load_module(): load_module: trying to load </usr/local Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [sr_module.c:357]: register_module(): register PV from: rtpproxy Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [cfg.lex:1724]: pp_define(): defining id: MOD_rtpproxy Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: ERROR: rtpproxy [rtpproxy.c:1707]: send_rtpp_command(): timeout waiting reply f Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:10.0.2.68:77 Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy <udp:10 Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: ERROR: rtpproxy [rtpproxy.c:1707]: send_rtpp_command(): timeout waiting reply f Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:10.0.2.68:77 Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP I run rtpproxy like this: /usr/local/rtpproxy-install-dir/bin/rtpproxy -l <public ip here> -s udp:10.0.2.68:7722 -F -d DBUG:LOG_LOCAL3 I have these linee in the kamailio.cfg file loadmodule "rtpproxy.so" # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:10.0.2.68:7722") I have tried with localhost, with 127.0.0.1 and get the same results ( I mean configure kamailio to contact rttpproxy with localhost instead of ip and at the same time running rtpproxy with the same reference. I can see the rtpproxy running/listening [root@ip-10-0-2-68 sbin]# netstat -lnp | grep rtpproxy udp 0 0 10.0.2.68:7722 0.0.0.0:* 3608/rtpproxy I can also see these lines: [root@ip-10-0-2-68 sbin]# netstat -np | grep 7722 udp 0 0 10.0.2.68:49772 10.0.2.68:7722 ESTABLISHED 3928/./kamailio udp 0 0 10.0.2.68:36097 10.0.2.68:7722 ESTABLISHED 3897/./kamailio udp 0 0 10.0.2.68:49158 10.0.2.68:7722 ESTABLISHED 3906/./kamailio udp 0 0 10.0.2.68:37129 10.0.2.68:7722 ESTABLISHED 3904/./kamailio udp 0 0 10.0.2.68:39052 10.0.2.68:7722 ESTABLISHED 3926/./kamailio udp 0 0 10.0.2.68:38161 10.0.2.68:7722 ESTABLISHED 3919/./kamailio udp 0 0 10.0.2.68:53144 10.0.2.68:7722 ESTABLISHED 3887/./kamailio udp 0 0 10.0.2.68:39721 10.0.2.68:7722 ESTABLISHED 3905/./kamailio udp 0 0 10.0.2.68:59563 10.0.2.68:7722 ESTABLISHED 3909/./kamailio udp 0 0 10.0.2.68:51884 10.0.2.68:7722 ESTABLISHED 3921/./kamailio udp 0 0 10.0.2.68:46902 10.0.2.68:7722 ESTABLISHED 3930/./kamailio udp 0 0 10.0.2.68:33981 10.0.2.68:7722 ESTABLISHED 3931/./kamailio udp 0 0 10.0.2.68:60221 10.0.2.68:7722 ESTABLISHED 3898/./kamailio udp 0 0 10.0.2.68:35651 10.0.2.68:7722 ESTABLISHED 3900/./kamailio udp 0 0 10.0.2.68:33105 10.0.2.68:7722 ESTABLISHED 3927/./kamailio udp 0 0 10.0.2.68:43349 10.0.2.68:7722 ESTABLISHED 3887/./kamailio udp 0 0 10.0.2.68:39259 10.0.2.68:7722 ESTABLISHED 3899/./kamailio udp 0 0 10.0.2.68:35170 10.0.2.68:7722 ESTABLISHED 3907/./kamailio udp 0 0 10.0.2.68:38760 10.0.2.68:7722 ESTABLISHED 3908/./kamailio In the rtpproxy I see that it is sent data and it replies too, I even got a wireshark capture showing that as well: Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3931_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3927_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3926_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3928_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" But somehow kamailio does not see it: [root@ip-10-0-2-68 sbin]# ./kamctl fifo nh_show_rtpp which: no greadlink in (/sbin:/bin:/usr/sbin:/usr/bin:/) which: no gdb in (/sbin:/bin:/usr/sbin:/usr/bin:/) udp:10.0.2.68:7722:: set=0 index:: 0 disabled:: 1 weight:: 1 recheck_ticks:: 53253287 I even when to the kamailio code and saw something that may have something to do with it but havent tried it yet:, in the implementation of the function "send_rtpp_command" I see that the socket created is using TCP. But I tried configuring kamailio/rtpproxy to use tcp instead but I also got the same results, kamailio seems to fail to acknoledgwe the replies from rtpproxy. I also have SELINUX disabled and the iptables are not running. I dont know what else to check, if somebody can shed some light into this I'd appreciate it. jp2014

1 1

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sr-users October 2014