sr-users May 2016

sr-users@lists.kamailio.org

86 participants
128 discussions

Kamailio Via: header issue
by Laura 27 May '16

27 May '16

Good morning list, I need to ask you a question about a problem we are fighting on our VoIP platform based on Kamailio 4.3.5. Our platform is made by more nodes over Europe countries, and we are encountering a problem with the size of the SIP package which, exceeds the physiological limit of about 1350 bytes. The real problem is caused from Via: Header (inside the SIP packet) which is added to each element of the system for transit on the call. >From what we have analyzed, the problem is … [View More]

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rtpproxy module question
by gmele 27 May '16

27 May '16

Hello, we are using an RTP Proxy from rtpproxy.org as media relay to establish communication between our mobile phones. Of course, we are using the kamailio rtpproxy module to modify the SDP payload and control the proxy. In our Kamailio configuration, we have 1 kamailio configured as Proxy and one kamailio configured as Registrar. So calls go through the Proxy and then to the Registrar who will update the SDP header and select an available rtp proxy. We have noticed that sometimes, the rtp … [View More]

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Kamailio not sending reply
by Jeremy Betts 26 May '16

26 May '16

Hello, I'm having an issue where kamailio is not sending the failure reply as I would expect when receiving calls from a certain provider. Kamailio sends the 403 response on calls from other providers. The big difference I see is that the "problem" provider is using compact headers. I am trying to force the 403 response to be sent for testing purposes but I just can't get it to reply to this one provider. I've included traces and configuration excerpts below. Any help is much … [View More]appreciated! Problem Call (No Reply Sent): U 63.79.178.192:5060 -> 184.171.164.100:5060 INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. Record-Route: <sip:63.79.178.192;lr>. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 INVITE. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144. Max-Forwards: 27. m: <sip:199.173.94.144:5060;transport=UDP>. k: 100rel, resource-priority, replaces. c: application/sdp. l: 235. P-Asserted-Identity: <sip:+17143258018@63.79.178.192;user=phone>. Privacy: none. . v=0. o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106. s=-. p=+1 6135555555. c=IN IP4 199.173.68.106. t=0 0. m=audio 55380 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=fmtp:18 annexb=no. # U 63.79.178.192:5060 -> 184.171.164.100:5060 ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 ACK. l: 0. Max-Forwards: 27. . Successful Call (Reply Sent): # U 64.136.173.31:5060 -> 184.171.164.100:5066 INVITE sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. Record-Route: <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp>. To: <sip:7146466334@184.171.164.100>. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Contact: <sip:17143258018@64.136.173.31:5060>. Supported: timer. Session-Expires: 1800;refresher=uac. Min-SE: 90. P-Asserted-Identity: <sip:17143258018@192.168.20.76>. Privacy: none. Expires: 120. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 274. . v=0. o=Sansay-VSXi 188 1 IN IP4 64.136.173.31. s=Session Controller. c=IN IP4 69.85.185.142. t=0 0. m=audio 37970 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 184.171.164.100:5066 -> 64.136.173.31:5060 SIP/2.0 403 DID Lookup Failed. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Server: x-Freevoice SIP Proxy 4.21. Content-Length: 0. . # U 64.136.173.31:5060 -> 184.171.164.100:5066 ACK sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 ACK. Max-Forwards: 70. Content-Length: 0. Kamailio Config: request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { setflag(FLT_ACCFAILED); if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(is_method("INVITE")||is_method("MESSAGE")){ route(ORIGINATE); #route(FROMAST); } route[ORIGINATE] { sl_send_reply("403", "DID Lookup Failed"); exit; } Jeremy Betts <https://www.youtube.com/user/FreevoiceLLC> <https://www.youtube.com/user/FreevoiceLLC> [View Less]

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webrtc tlsv1 alert access denied
by Arsen 26 May '16

26 May '16

Hi guys! I am trying to configure kamailio with WSS. We have trusted certificate installed SIP over TCP/TLS works fine. But when I try WSS I got error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL routines:SSL3_READ_BYTES:tlsv1 alert access denied ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading Before above error it was 'bad certificate', so I have imported CA in the firefox and now I get these errors.. I have tried … [View More]

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SRV and dialog stickiness
by Alex Balashov 26 May '16

26 May '16

Hello, Sorry if this is a tired, worn question, but I've not dealt much with Kamailio's SRV support before: If a registrant has a contact binding whose domain component is subject to an SRV lookup with load-balanced or weighted entries, how does one solve the problem of ensuring that subsequent in-dialog requests go to the same host as the initial INVITE? Does Kamailio offer some facility for doing this? Is it somehow accommodated by SIP? -- Alex -- Alex Balashov | Principal | … [View More]

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rtpengine and SAVPF
by Ján Füri 26 May '16

26 May '16

Hi all, Im struggling with incoming calls to webrtc clients. I am trying to change INVITE's SDP using rtpengine from RTP/AVP or RTP/SAVP to RTP/SAVPF towards my webrtc clients. This works well. But if I send INVITE with SDP that includes RTP/AVP and RTP/SAVP as well, rtpengine creates incorrect SDP. It includes twice m=audio RTP/SAVPF and my WebRTC client does not like this INVITE and forbiddeds it by 488 Not Acceptable Here. Example of my incorrect INVITE : Content-Type: application/… [View More]

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Sequential calls messes up presence with pua-dialog info
by Ali Pey 25 May '16

25 May '16

Hello, We are using pua-dialoginfo and presence modules for BLF. When the first call comes in or goes out, every thing works great; however, a second call to the same key/extension (e.g a call waiting which is not answered) would turn off the key even though the original call/dialog is still active. Has anyone had experienced such problem? How can I fix this? Any constructive suggestion would be appreciated. Regards, Ali Pey

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nh_reload_rtpp not available
by david 25 May '16

25 May '16

dear all i'm using kamailio 4.4.1 and i'm not seeing nh_reload_rtpp as a fifo exported function i'm trying to setup the dburl for the rtpengine sets using db_text but when i do kamctl fifo which i see all the commands of the module except the reload_rtpp nh_enable_rtpp nh_show_rtpp nh_ping_rtpp nh_show_hash_total /usr/local/kamailio/sbin/kamctl fifo nh_reload_rtpp 500 command 'nh_reload_rtpp' not available these are the config lines i have for that #!define DBURLRTP "text:///var/log/… [View More]

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doubt using rate limit
by david escartin 25 May '16

25 May '16

Hello Daniel i had the call rate for some minutes actually. maybe 4-5min and all the time the behaviour was like that. it's true that in that kamailio there wa no any other traffic, only those test calls i set with a sipp to simulate the load best regards david Hello, for how long you got this 20req/second? The module is not counting like continuous time, but more like traffic on each second, iirc. Cheers, Daniel On 23/05/16 11:24, david escartin wrote: > Hello all > > i … [View More]

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Kamailio for sip routing architecture
by Loic Chabert 25 May '16

25 May '16

Hello Kamailio users, I'm working on a sip infrastructure, with opensource technologies, and of course Kamailio will run on my new infrastructure :) Kamailio will be my central routing engine, between SBC (running on asterisk), and somes boxes (asterisk too) where customers will be registred. Please find below my future infrastructure: ---------- -------------- -------- ---------------- | SBC1 | - - - - - | - - - -| kamailio1 |- - - - - | - - - - - … [View More]

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sr-users May 2016