- sr-users - kamailio.org

kamailio / rtpproxy - remove codec
by Oliver Roth 08 Apr '14

08 Apr '14

Hi all We use kamailio 3.3.7 and rtpproxy for enduser call-termination. In case of a fax call, we get an invite from our carrier for codecs G711a/u and T38. As our termination carrier does not support T38 and because the invite contains G711 and T38 we get back error 488. How is it possible to remove the whole T38 part of this invite? We tried sdp_remove_codecs_by_name(list) without success - what "name" should we use for T38? [T38, t38, t.38, T.38, ...] sdp_remove_line_by_prefix(string) sdp_remove_media(type) None of these functions did really work - best was the last one with type=image but then the sip header is malformed. As we saw with Kamailio version 4.1.x there are a lot of new functions within sdpops. Would an upgrade help? So basically the question is: How to remove the t38 part of the fax invite? (see attachment) KR, Oli [cid:image001.png@01CF523C.988433F0]

2 3

SIP 480 - Service unvailable
by Amogh meshram 07 Apr '14

07 Apr '14

Hello everyone, >From today i started getting SIP 480 error when making outbound call. I checked out with a different voip provider and got SIP 500 error. Internet calling is working fine. The service (outbound calls) was working fine till yesterday. I'm a newbie to kamailio and voip, so need your help. Please guide me to resolve this error. Thanks in advance.

2 11

SIP client TCP behaviour
by Alex Balashov 07 Apr '14

07 Apr '14

Hello, I don't know a lot about SIP with TCP, so I thought I'd ask for some opinions here. I've been testing Kamailio with TCP against the Android SIP client (4.4.2) and have found, to my annoyance, that it sends a RURI with a transport attribute in its INVITEs: INVITE sip:14045551212@sip.evaristesys.com;transport=tcp My upstream gateways are all UDP, so this causes Kamailio to attempt to contact them all via TCP (and fail). I know how to bend the transport with Kamailio, and I can strip this attribute. My question is more methodological: is this "correct" for a client to do? In my opinion, the answer is no. The client should not be so presumptuous. It should specify the transport in its Contact, to indicate how it wants to be reached, and leave other decisions about transport to the UAS. But is there something I am perhaps missing? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 4

TLS connection with short lifetime - how to handle it?
by Dmytro Bogovych 07 Apr '14

07 Apr '14

Greetings. I have the next problem: iOS based clients connect via TLS to kamailio server. They run mostly in background mode - it means connection refresh interval is ~10 minutes. Some of clients reside behind paranoidal routers which considers such idle connections as lost and closes them. I see only way to resolve it is to send OPTIONS/NOTIFY/CRLF from kamailio to clients. Tried to use nat_traversal module by calling nat_keepalive() to every REGISTER message. But there is no incoming keepalive (OPTIONS or NOTIFY) on client side. Is there another way to make "heartbeat" in SIP/TLS from server side? Thank you.

1 0

One-way Relay through RTPproxy
by Nandini madhu 07 Apr '14

07 Apr '14

Dear All, I am running Kamailio (4.0) and RTPproxy server (1.2.1) server in my set-up. When i call from Client A to B , the RTP packets are proxying trhrough RTPproxy service to other end client, but no flow from client B to client A. With this calls Are unsuccessful. I traced a SIP capture (please find in the attachment) and analysed it, as in there only one client 's (A) IP address is changed to RTPproxy IP address in SDP body. What can be do to resolve this issue ? Please anybody help me And also find the attachment for my kamailio config file, and suggest me or point me about any misplacement of NAT traversal function call in the script. Please consider this and help me out as i am newbie in scripting. Regards, Nandini

2 2

Show Domain Name in Siremis
by aawaise 07 Apr '14

07 Apr '14

Hello, I have a Kamailio-3.1.1 running with its DB on MySQL running on a seperate machine. Siremis is installed for interaction with DB. In web interface of Siremis, domain name of users is not shown. How can I show that ?? Like my domain name is "defg" and user is "abc". It shows online users as "abc@HostNameOfUser" instead of "abc(a)defg.com". Secondly does Siremis read information from Kamailio server or directly from DB server ?? Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp12653… Sent from the Users mailing list archive at Nabble.com.

1 0

Restricting Kamailio to single login
by MrIhaveAnOpinionOnEverything 05 Apr '14

05 Apr '14

Hi Everyone: I am a newbie in kamailio. Basically I am looking for a way to restrict the registration or login in kamailio per user to just one. The experience I want to setup is to unregister an older register/login to kamailio by a new login/register after the register command has already passed the authentication leg. Is there an example for this? Thanks. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

3 2

Kamailio Freeswitch
by MrIhaveAnOpinionOnEverything 05 Apr '14

05 Apr '14

To whom it may concern: We are configuring a SIP platform with Kamailio and Freeswitch with this setup: UAC 1 ==> Kamailio ==> Freeswitch UAC 2 <== Kamailio <== Freeswitch We followed the instructions in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc We are using Kamailio on kamailio 4.0.3 and Freeswitch FreeSWITCH Version 1.5.6b Everything is working fine with the default setting. But when we configured TLS what we noticed some errors. UAC configured with no TLS calling a UAC configured with TLS works ok. UAC configured with TLS calling a UAC configured with TLS gets disconnected 15-30 seconds after answering the call. UAC configured with TLS calling a UAC configured with no TLS gets disconnected 15-30 seconds after answering the call. I cant seem to find any error in the logs or in the ngrep. Can anyone assist me regarding this issue. Good day. -- "When I look at you I see two people, ther person you are and the person you are supposed to be. Someday these two people will meet. And when they do, they will achieve great things" - Gene Hackman, The Replacements Catch up on me and check my BLOG at http://mrihaveanopiniononeverything.blogspot.com

2 3

Problem on dialog and reInvite
by pars3c 05 Apr '14

05 Apr '14

Hi I've a problem of this type: On the first INVITE is all ok, insert record route with cookie of the dialog: Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.1661> The I recieive a 422 for the session timer too small . I resend the new invite with correct expire but with a new cookie dialog value Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661> Then the call is connected (Record-Route: <sip:x.x.x.x;lr=on;ftag=as0f04b1b4;didsw=007.2661>). The there is a reInvite for the session timer , the script enter in the routine WITHINDLG . At this point the dialog seems to be lost. The accounting is made with dialog variable and in the 2 records INVITE and BYE , the fields corresponding to these values are missing. In all the other calls without the reInvite for session timers, the dialog is found and the variables are present. I don't understood what is problem ? Thanks for the help

1 1

Kamailio 4.1 and Cisco 79xx Integration Work
by Jonathan Hunter 04 Apr '14

04 Apr '14

Hi Guys, I'm doing some work, with Legacy Cisco phones, its a long story but I need to manipulate signaling due to firmware limitations. I am looking to implement a workaround, whereby I need to modify an ACK, so that it sends the RURI is sent to port 5060, and the destination Port at the IP layer is unchanged, I can do this with static values as below; sethostport("1.2.3.4:5060"); t_relay("1.2.3.4","38596"); In terms of doing this for multiple sites I need to use pseudo variables, so I wondered how this could be achieved, as t_relay or forward dont support dynamic values to this point, or do they? Any comments would be great. Many thanks Jon

2 2

Record-Route headers all over the place
by Phillman25 Kyriacou 04 Apr '14

04 Apr '14

Hi im receiving the following INVITE on my system, you can see that the Record-Route headers are all over the place, is there a way to group the Record-Route headers together followed by the via headers grouped together? INVITE sip:22000017@81.21.42.184;uniq=DBFF66459C85D6171B30016D2422B SIP/2.0 Record-Route: <sip:81.21.32.84;lr=on;ftag=as4c938f35;did=6da.a472> Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35> Via: SIP/2.0/UDP 81.21.32.84;branch=z9hG4bK6a96.dc3413ac122b4d4b8fd61bf19d95b1cf.0 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bK6a96.c81d2c2ad30db0820f05268ea67896f7.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-477861-22030313-22000017-74007 Record-Route: <sip:81.21.38.5;lr=on;ftag=as4c938f35> Record-Route: <sip:81.21.38.34;lr=on;ftag=as4c938f35;did=6da.0ff1> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bK6a96.5d12e421943fbaec75df9a6a8091d336.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7e570efe;rport=5060 Max-Forwards: 16 From: <sip:22030313@192.168.10.189>;tag=as4c938f35 To: <sip:22000017@81.21.38.34> Contact: <sip:22030313@192.168.10.189:5060> Call-ID: 74d7379b710c82f956d7cb31615c2581@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(11.7.0) Date: Fri, 04 Apr 2014 11:33:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 257 P-hint: outbound P-hint: outbound v=0 o=root 1396887739 1396887739 IN IP4 81.21.38.36 s=Asterisk PBX 11.7.0 c=IN IP4 81.21.38.36 t=0 0 m=audio 42746 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv thanks in advance! Phillip

2 1

What going on this SDP
by jaflong jaflong 04 Apr '14

04 Apr '14

Hi List, Can anyone help me understand why this is getting rejected Please note the specific message further dow the log. "Failed to parse SessionDescription. Failed to parse audio codecs correctly" This is on Chrome. On Firefox There is a further message in the console "Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: Transport protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: Invalid port format (17296) specified for transport protocol (Unsupported), parse failed." JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414 JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0 jssip-0.3.0.js:3369 JsSIP | TRANSPORT | sending WebSocket message: INVITE sip:9822@10.1.1.101 SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581 Max-Forwards: 69 To: <sip:9822@10.1.1.101> From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Contact: <sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INVITE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 1744 v=0 o=- 3746191339358890844 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 10.10.10.63 a=rtcp:65223 IN IP4 10.10.10.63 a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0 a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0 a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0 a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0 a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0 a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0 a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0 a=ice-ufrag:Dgp8HIJdmr1lFPCQ a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9 a=ice-options:google-ice a=fingerprint:sha-256 C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 768e2e47-bc86-473d-bc2c-6e2340ace772 a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8 a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772 jssip-0.3.0.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63 To: <sip:9822@10.1.1.101> From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Server: DXI WebRTC Content-Length: 0 Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455 req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101 out_uri=sip:9822@10.10.10.111:5443 via_cnt==1" jssip-0.3.0.js:670 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WSS flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov To: <sip:9822@10.1.1.101>;tag=as06b3db08 Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 INVITE Server: Easycall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:9822@10.10.10.111:5443;transport=TLS> Content-Type: application/sdp Content-Length: 801 v=0 o=root 431209641 431209641 IN IP4 10.10.10.111 s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.10.10.111 t=0 0 m=audio 30490 UDP/TLS/RTP/SAVPF 0 126 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138 a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092 a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv jssip-0.3.0.js:670 Failed to parse SessionDescription. Failed to parse audio codecs correctly. jssip-0.3.0.js:4512 JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523 JsSIP | TRANSPORT | sending WebSocket message: ACK sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640 Max-Forwards: 69 To: <sip:9822@10.1.1.101>;tag=as06b3db08 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7435 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 jssip-0.3.0.js:519 JsSIP | TRANSPORT | sending WebSocket message: BYE sip:9822@10.10.10.111:5443;transport=tls SIP/2.0 Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766 Max-Forwards: 69 To: <sip:9822@10.1.1.101>;tag=as06b3db08 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov Call-ID: 43oclsi0sva6n347bk5c CSeq: 7436 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.0 Content-Length: 0 jssip-0.3.0.js:519 JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov jssip-0.3.0.js:4193 JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392 JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted jssip-0.3.0.js:2543 JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WSS flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766 From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov To: <sip:9822@10.1.1.101>;tag=as06b3db08 Call-ID: 43oclsi0sva6n347bk5c CSeq: 7436 BYE Server: Easycall Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0

4 6

Load new module in Kamailio
by Patrik Kristel 04 Apr '14

04 Apr '14

Hello all, I would like to ask, how can I load new module in Kamailio 4.1.2? Actually, I have an issue, when I tried to compile my kamaiio.cfg I've got error: root@kamailio:/usr/local/# kamailio -c kamailio.cfg loading modules under /usr/local/lib64/kamailio/modules/ 0(25392) ERROR: <core> [sr_module.c:587]: load_module(): ERROR: load_module: could not find module <websocket> in </usr/local/lib64/kamailio/modules/> 0(25392) : <core> [cfg.y:3408]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 323, column 12-25: failed to load module 0(25392) ERROR: <core> [cfg.y:3272]: yyparse(): cfg. parser: failed to find command ws_handle_handshake 0(25392) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1083, column 27: unknown command, missing loadmodule? ERROR: bad config file (2 errors) when I look physically to the /usr/local/lib64/kamailio/modules/ there is some modules, but websocket.so is missing. So, how can I get and load module in Kamailio? Thank you for help! BR, Patrik Kristel

2 1

Crash on REGISTER
by Igor Potjevlesch 03 Apr '14

03 Apr '14

Hello, I experienced an issue yesterday. I move a customer to Kamailio 4.0.4 (previously he was on an old SER instance without issue). He uses the same SIP account for connecting two IPBX (same platform, same firmware). When the second REGISTER came on Kamailio and, may be 500ms/1s after (I have sufficient time to see 200 OK (2 bindings) on Wireshark), Kamailio crash with coredump generated. I know that I must analyse the coredump but I hope that could be a known issue. In addition, when I try to run "gdb" on the core files, I got the following: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 256 -M 64'. Program terminated with signal 11, Segmentation fault. #0 0x00000030f2230f30 in escape_string_for_mysql () from /usr/lib64/mysql/libmysqlclient.so.16 Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.9-33.el6.x86_64 libcom_err-1.41.12-12.el6.x86_64 libconfuse-2.6-2.el6.rf.x86_64 libselinux-2.0.94-5.3.el6.x86_64 mysql-libs-5.1.61-4.el6.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64 openssl-1.0.0-20.el6_2.5.x86_64 zlib-1.2.3-27.el6.x86_64 I guess that I'm missing packages to troubleshoot. Many thanks. Regards, Igor.

3 10

NAT Traversal issue
by Wingsravi R 03 Apr '14

03 Apr '14

Dear Kamailio'ns, I am working on Kamailio server (V 4.1.2) with RTPproxy (1.2.1) integrated, in a standalone intranet infrastructure (no any connection with internet). I dont have any NAT settings in my network set-up. even though i will not get Audio/video calls through some times. So in that concern I have installed RTPproxy, now all the audio/Video calls are fine (with some Pixelled). I have the Following Kamailio configuration script, in which it suppose to invoke RTPproxy service when the SIP clients behind NAT. But every time when i do Audio/Video calls, they are proxying through RTPproxy server only. I analysed SIP captures of Audio/video call, i didnt found any IP/port changes in the whole SIP session and with this i assumed that there is no NAT issue in my Network. But why all the Audio/Video calls are proxying through RTPproxy everytime ? Is there any Wrong placement of function call in Kamailio configuration script (below) ? #----------------------------------------------------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@192.168.2.52") modparam("nathelper", "sipping_method", "INFO") # ----- NAT_traversal ----- modparam("nat_traversal", "keepalive_interval", 60) modparam("nat_traversal", "keepalive_method", "NOTIFY") modparam("nat_traversal", "keepalive_state_file", "/var/run/kamailio/keepalive_state") # ----- params needed for NAT traversal in other modules ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", 6) #!endif #Routing Script # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; #if (msg:len > max_len) { if (msg:len >= 8192 ) { sl_send_reply("513", "Message too big"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #if (method!="REGISTER") { if (!method=="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { # unforce_rtp_proxy(); rtpproxy_answer(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("192.168.2.52","subscriber")) { # proxy_challenge("","0"); proxy_challenge("192.168.2.52", "0"); # break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; rtpproxy_offer("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # if (uri!=myself) { if (!uri==myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { rtpproxy_answer();; }; sl_reply_error(); }; } # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- route[2] { if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("192.168.2.52","subscriber")) { www_challenge("192.168.2.52","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { if (!proxy_authorize("192.168.2.52","subscriber")) { proxy_challenge("192.168.2.52","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- route[4] { if (isflagset(6)) { force_rport(); fix_nated_contact(); # force_rtp_proxy(); rtpproxy_offer(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { # force_rtp_proxy(); rtpproxy_offer(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } Please find the Attachment for Tcpdump based Video call Sip capture for your better Understanding. PS: Both Kamailio and RTPproxy are running on same IP (host), i.e 192.168.2.52. Please anybody help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

3 2

when dialing another extension, instead of ringing for 30 sec, vmail kicks in right away
by mark li 03 Apr '14

03 Apr '14

I'm using kamailio as a sip proxy and registrar, and freeswitch for all media services like vmail, conference calls, ivrs etc. I just set up vmail ... and it seems to be working. Except that it appears that I've broken something else because now when I call extB from extA, instead of ringing first for a little while and then having the vmail kick in afterwards, vmail kicks in right away. someone mentioned that it could be a forking problem. I'm afraid I don't really understand what this means. just wondering if you can point me in the right direction list, as far as what I should be looking for in the kamailio.cfg file... Thanks.

2 2

error messages in kam log whenever i do anything with my phone
by mark li 02 Apr '14

02 Apr '14

Hi there. I've noticed that no matter what i do with my test phone (call voicemail, call another extension etc) I get error messages like the following in my log file: Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=553 a=26 n=xlog Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=560 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=557 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=563 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=560 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=567 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=563 a=25 n=is_method Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=570 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19400]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=567 a=24 n=t_relay Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1064 a=25 n=xdbg Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1067 a=16 n=if Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1071 a=5 n=route Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=854 a=2 n=return Apr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1088 a=16 n=ifApr 2 14:31:07 jl-raspberrypi /usr/sbin/kamailio[19397]: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=1073 a=24 It doesn't seem to impact the operation of the phone / dialplans etc. I recall reading somewhere that it might have to do with caller ids or location. I'm trying to google to see if I can find the article the made references to similar error messages. But in the meantime, I thought I'd check the list to see if you could point me in the right direction.

2 1

Re: [SR-Users] Regarding Websocket module
by Juha Heinanen 02 Apr '14

02 Apr '14

Premchandiran writes: > Not able to find libunistring-dev rpm for Linux 2.6.18-274.el5. almost 4 > hours I am searching for the rpm. switch to debian, -- juha

2 1

Understunding the routing logic with WebSockets
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Hi experts, Using websockets, I am facing a behavior I can't manage explaining : on reception of an initial REGISTER (over websocket), it seems that the routing logic in the request_route method is not invoked Extract of my Kamailio.cfg to explain =================================================================== Request_route { xlog("L_DBG", "Entering request_route for $rm") ... Onreply_route[REGISTER_reply] { If (t_check_status("200")) xlog("L_DBG", "REGISTER OK") Event_route[xhttp:request] { xlog("L_DBG", "XHTTP Request received") ws_handle_hanshake(); In my logs, I only see > XHTTP Request received > REGISTER OK No trace of "Entering request route for REGISTER" ! whereas for a subsequent INVITE, I see the trace Any explanation ? because of this, the fix_nated_register() is not called, and I suspect this causes the troubles I meet trying to make call with websockets Olivier DURECU Bell Labs / IP Platform Research Villarceaux Center Tel: +33(0) 130772759 GSM: +33(0) 683776482

1 0

R: Reply route and variable
by pars3c 02 Apr '14

02 Apr '14

Hello, Probably i dont see the dialog variable in the onreply route , because I see on the trace ip that sometime on the reInvite the proxy dont put on the record route the dialog cookie. On the first invite , its ok , but on every reinvite (we have sip session timer) there isnt. Its correct ? Thanks Da: pars3c [mailto:pars3c@gmail.com] Inviato: martedì 1 aprile 2014 17:32 A: 'Kamailio (SER) - Users Mailing List' Oggetto: Reply route and variable Hi , i have a question for all : in the onreply_route is visible a dialog variable ($dlg_var(ring_time)) or a avp variable ? Can I set them , on this kind of route ? On the event route (dialog start) they are all null. I try to set them, but some time (on reInvite) this variables are not setted. Why? Thanks to all

1 0

Re: [SR-Users] Regarding Websocket module
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Have a look at http://rpm.pbone.net/index.php3/stat/3/limit/1/srodzaj/1/dl/40/search/libun…<http://rpm.pbone.net/index.php3/stat/3/limit/1/srodzaj/1/dl/40/search/libun…> and search for your OS version through the pages Olivier De : sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:17 À : 'Kamailio (SER) - Users Mailing List' Objet : Re: [SR-Users] Regarding Websocket module Hi Olivier, Not able to find libunistring-dev rpm for Linux 2.6.18-274.el5. almost 4 hours I am searching for the rpm. Thanks, Prem From: sr-users-bounces(a)lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org> [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of DURECU, OLIVIER (OLIVIER) Sent: Wednesday, April 02, 2014 12:39 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Regarding Websocket module Hi, You may be missing a library. On Ubunutu/debian, this is the libunistring-dev package which provides the required library This is documented on the module description (modules/websocket/README) Olivier De : sr-users-bounces(a)lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org> [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:00 À : sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Objet : [SR-Users] Regarding Websocket module Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

1 0

Re: [SR-Users] Regarding Websocket module
by DURECU, OLIVIER (OLIVIER) 02 Apr '14

02 Apr '14

Hi, You may be missing a library. On Ubunutu/debian, this is the libunistring-dev package which provides the required library This is documented on the module description (modules/websocket/README) Olivier De : sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] De la part de Premchandiran Envoyé : mercredi 2 avril 2014 09:00 À : sr-users(a)lists.sip-router.org Objet : [SR-Users] Regarding Websocket module Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

2 1

Regarding Websocket module
by Premchandiran 02 Apr '14

02 Apr '14

Hi All, I am using kamailio-4.1.2 and os is Linux 2.6.18-274.el5. I am getting below error though websocket.so is present in /usr/local/lib64/modules/ load_module(): ERROR: load_module: could not open module </usr/local/lib64/kamailio/modules/websocket.so>: libunistring.so.0: cannot open shared object file: No such file or directory may I know what might be the cause of this error. Regards Prem

1 0

Dimension a kamailio server
by Rizwan Khan 02 Apr '14

02 Apr '14

HI Guys, can you refer me to some resources which help me dimension the hardware for a setup. Total users will be 20,000. What percentage should i assume for concurrent audio and video calls? What is the standard practice? How does this all map to the cpu, ram and storage etc. given that I will be storing the cdrs as well. In the futute this setup should be scalable so that it should support 200,000+ users within a span of 6-8 months. What would be the ideal architecture? Thanks, Rizwan Khan

4 9

loose_route() problem
by Juha Heinanen 01 Apr '14

01 Apr '14

my proxy is listening at udp/tcp ip:port 192.98.102.30:5060 and my script has this for in-dialog requests: route [IN_DIALOG_REQUEST] { # handle in-dialog request if (!loose_route()) { xlog("L_INFO", "Loose route failed on $hdr(route)\n"); ... exit; }; xlog("L_INFO", "destination uri after loose_route: <$du>\n"); if i send to my proxy with sipp this in-dialog bye: <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="In-dialog BYE"> <send> <![CDATA[ BYE sip:test@192.98.102.30:53846;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 192.98.101.40;branch=z9hG4bK63f.e9d25102.0 Via: SIP/2.0/UDP 192.98.101.157:5080;branch=z9hG4bK6955a179;rport=5080 Route: <sip:192.98.102.30;r2=on;lr;n1>,<sip:192.98.102.30;transport=tcp;r2=on;lr;n1> Max-Forwards: 16 From: <sip:jh@test.tutpro.com>;tag=as0cb9dd4f To: "Test" <sip:test@test.tutpro.com>;tag=3feea37d Call-ID: d745a79b66919309ba2ef68fde0ca13e@0:0:0:0:0:0:0:0 CSeq: 102 BYE Content-Length: 0 ]]> </send> </scenario> i correctly get to syslog: Apr 1 20:47:28 siika /usr/sbin/sip-proxy[9583]: INFO: destination uri after loose_route: <<null>> that is, the proxy removed both route uris that pointed to itself from the route set leaving empty $du. however, if i send to my proxy with sipp this in-dialog bye (where the only difference to the previous one is request uri that now contains alias param): <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="In-dialog BYE"> <send> <![CDATA[ BYE sip:test@192.98.102.30:5060;alias=192.98.102.30~53846~2;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 192.98.101.40;branch=z9hG4bK63f.e9d25102.0 Via: SIP/2.0/UDP 192.98.101.157:5080;branch=z9hG4bK6955a179;rport=5080 Route: <sip:192.98.102.30;r2=on;lr;n1>,<sip:192.98.102.30;transport=tcp;r2=on;lr;n1> Max-Forwards: 16 From: <sip:jh@test.tutpro.com>;tag=as0cb9dd4f To: "Test" <sip:test@test.tutpro.com>;tag=3feea37d Call-ID: d745a79b66919309ba2ef68fde0ca13e@0:0:0:0:0:0:0:0 CSeq: 102 BYE Content-Length: 0 ]]> </send> </scenario> i get to syslog: Apr 1 20:51:20 siika /usr/sbin/sip-proxy[9540]: INFO: destination uri after loose_route: <sip:192.98.102.30;transport=tcp;r2=on;lr;n1> can someone explain, how it is possible that when request uri changes, loose_route() does not anymore consume both route uris that point to the proxy itself? could this be a bug in loose_route()? -- juha

1 1

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sr-users