sr-users October 2014

sr-users@lists.kamailio.org

101 participants
188 discussions

append_hf on kamctl fifo dlg_bridge destroys ACK
by Kristian F. Høgh 31 Oct '14

31 Oct '14

Hi, Tested on 4.0.6 (+contact header REFER patch) and trunk. When appending header on originating INVITE, the ACK from kamailio gets messed up. I generate call, using myself as proxy kamctl fifo dlg_bridge sip:201@[domain] sip:202@[domain] sip:[kamailio-ip]:5060 In route[tm:local-request], i use append_hf("P-Asserted-Identity:...") If I remove append_hf, the ACK looks OK. Regards, Kristian Høgh Trace on local interface: INVITE sip:201@[domain] SIP/2.0 Via: SIP/2.0/UDP [kamailio-ip];branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 To: <sip:201@[domain]> From: <sip:controller@kamailio.org>;tag=760f7752be879bf1a1346b149d21b85f-8c89 CSeq: 10 INVITE Call-ID: 0b4f5d725846cc40-12109@[kamailio-ip] Max-Forwards: 70 Content-Length: 147 User-Agent: kamailio Contact: <sip:controller@kamailio.org:5060> Content-Type: application/sdp P-Asserted-Identity: <sip:201@[domain]> <snip SDP> SIP/2.0 200 OK Via: SIP/2.0/UDP [kamailio-ip];rport=5060;branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 Record-Route: <sip:[kamailio-ip];lr;ftag=760f7752be879bf1a1346b149d21b85f-8c89;vsf=AAAAAFFfXzREVlRVXUR1WU8dAxFCHAFHQgYCbC5kaw--;did=abe.bd12> Contact: <sip:201@[peer-ip]:43017;rinstance=50924dd7a7927b2f;transport=UDP> To: <sip:201@[domain]>;tag=594df15d From: <sip:controller@kamailio.org>;tag=760f7752be879bf1a1346b149d21b85f-8c89 Call-ID: 0b4f5d725846cc40-12109@[kamailio-ip] CSeq: 10 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.0.19649 r19608 Allow-Events: presence, kpml Content-Length: 157 <snip SDP> ACK sip:201@[peer-ip]:43017;rinstance=50924dd7a7927b2f;transport=UDP SIP/2.0 Via: SIP/2.0/UDP [kamailio-ip];branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 \0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00P\0xeeu\0x00\0x00\0x00\0x00\0x00$\0x12v\0x00\0x00\0x00\0x00\0x00\0xbb\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0xf0\0xf0\0xf0\0xf0\0x00\0x00\0x00\0x00\0x16\0x0075\0x13\0x01\0x00\0x00X\0xfc\0xc2Vk\0x7f\0x00\0x00\0x00\0xfd\0xc2Vk\0x7f\0x00\0x00\0x0a\0x00\0x00\0x00f-8c\0x0b\0xfd\0xc2Vomain\0x00<domain>\0x009@178X\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00To: <sip:201@[domain]>;tag=594df15d k\0x7f\0x00\0x00\0x17\0x00\0x00\0x00 ACK Route: <sip:[kamailio-ip];lr;ftag=760f7752be879bf1a1346b149d21b85f-8c89;vsf=AAAAAFFfXzREVlRVXUR1WU8dAxFCHAFHQgYCbC5kaw--;did=abe.bd12> User-Agent: kamailio Content-Length: 0 REFER ...

2 4

Susceptibility to POODLE Vulnerability?
by Rainer Piper 31 Oct '14

31 Oct '14

Hi all, is it possible to add in http://kamailio.org/docs/modules/4.2.x/modules/tls.html under the line > 9.1. |tls_method| (string) ... ... If rfc3261 conformance is desired, TLSv1 must be used. For compatibility with older clients SSLv23 is a good option. *Example 1.3. Set |tls_method| parameter* ... modparam("tls", "tls_method", "TLSv1") ... < !!! *a warning **that the use of SSLv3 **susceptibility to POODLE Vulnerability* !!! -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rainer@sip.soho-piper.de:5072 (pjsip-test) XMPP: rainer(a)xmpp.soho-piper.de

4 6

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

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sr-users October 2014