[SR-Users] Want dialog.default_timeout NOT to reset on subseq. req

Hi! I'd like to restrict max call duration. Currently we have something that partially works: dialog.default_timeout. But, as described in "How it works" section http://www.kamailio.org/docs/modules/devel/modules/dialog.html#idp2029360 "The dialog timeout is reset each time a sequential request is

Re: [SR-Users] dialog keepalive timeout - which parameter rules it?

2015-07-21 10:09 GMT+03:00 Daniel-Constantin Mierla : > Hello, > > it is fr_timer from tm module. Thank you very much for your answer. -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@

[SR-Users] dialog keepalive timeout - which parameter rules it?

alive request sent by Kamailio within dialog triggers dialog termination. Thanks a lot. -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/ma

Re: [SR-Users] Double Record-Route to subscriber with multiple locations on different SIP transports

Call-ID: RCYQTYzQlO Max-Forwards: 69 Supported: outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 1714 Contact: ;+sip.instance="" User-Agent: LinphoneAndroid/2.4.1-19-

Re: [SR-Users] Double Record-Route to subscriber with multiple locations on different SIP transports

6:12 webrtcloadtest kamailio: : [cfg.y:3432]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 29, column 1-6: ip address, interface name or hostname expected Please see that I already use advertised_address directive, but anyway I have that

Re: [SR-Users] Double Record-Route to subscriber with multiple locations on different SIP transports

nd let you know the results. There's new RR-related issue: now tested it in different setup, and record_route_advertised_address() doesn't write "transport=" attribute when the call goes from websocket to TLS. So Linphone sends BYE via UDP again. -- Andrey Utkin

Re: [SR-Users] Double Record-Route to subscriber with multiple locations on different SIP transports

ute_advertised_address() because other record_route functions put internal IP to the header (the server is behind Amazon firewall). We call record_route_advertised_address() just one time, but in two places, meant to handle different cases. -- Andrey Utkin

[SR-Users] Double Record-Route to subscriber with multiple locations on different SIP transports

ecord-route Could anyone explain what would be the most correct solution? What is intended behaviour of branch_route? Is it omitted from execution on "primary" branch of call? Is there any parameter which could help? Thanks in advance. -- Andrey Utkin _

Re: [SR-Users] dialog: use keepalive & timeout features together

2015-07-08 15:22 GMT+03:00 Andrey Utkin : > 2015-07-07 10:07 GMT+03:00 Daniel-Constantin Mierla : >> if there is no response to keepalive, the call should be terminated in >> like 10 seconds. > > Thank you for quick reply Daniel. > Can this be different for the case of

Re: [SR-Users] dialog: use keepalive & timeout features together

2015-07-07 10:07 GMT+03:00 Daniel-Constantin Mierla : > if there is no response to keepalive, the call should be terminated in > like 10 seconds. Thank you for quick reply Daniel. Can this be different for the case of TLS connections? I guess this can matter. -- Andrey

[SR-Users] dialog: use keepalive & timeout features together

is better. Thanks in advance. -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

led PDU]". Any review is greatly appreciated! -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

ead of ngrep to have more details and confidence? Also I guess you mean this to be an issue of Linux kernel on any side, or possibly of routing hardware somewhere in the route? -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) -

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

rting TCP, I'm afraid this can be considered Kamailio issue (honestly, I still don't quite believe as I percept Kamailio as robust and stable software). Any review and comment helps. -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER)

Re: [SR-Users] How to disable retransmits via TCP connection? etc.

t; partial sip packets is ok as long as the receiving party doesn't > complain of broken/incomplete sip packet. I accept the chance that I misinterpret the output of ngrep. So I need to collect more info, like the traffic from client network. Than

[SR-Users] How to disable retransmits via TCP connection? etc.

e will check how it works with TLS transport a bit later when there's technical possibility. I have two questions: 1. How would I completely disable retransmissions to TCP connections? 2. Any ideas what can be the reason for this issue? Retransmissions by

Re: [SR-Users] lookup() to fork execution of routing script?

_NAME. > > http://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_on_branch > > Hope this helps. Thank you, this (+ always setting explicitly the needed RTP profile in rtpengine_manage() + updating rtpengine to have a fix of rtpengine's github issue #103) have solv

[SR-Users] lookup() to fork execution of routing script?

o all available user locations. So we need an advice how to go. Is there a way to implement what we need at scripting level, or should this feature be implemented in Kamailio code (or is it already there?). Thanks in advance for any help. -- Andrey

Re: [SR-Users] Amazon VPS, long UDP packets are seen by sniffer, but don't reach application

s. I think we can live with it so far, as amount of traffic is not that big, and it is also effectively compressed on the fly as it is plain text. -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lis

Re: [SR-Users] Amazon VPS, long UDP packets are seen by sniffer, but don't reach application

nyway, will try with tunnel (which is, howevre, a workaround, not a solution). -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Amazon VPS, long UDP packets are seen by sniffer, but don't reach application

n VPS. I am concerned with this issue because rtpengine software has UDP interface. So on Amazon hosts this interface works only within localhost, and I cannot distribute software to different nodes. Any thoughts? What's wrong, how to fix? -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] [BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)

www-client/google-chrome-unstable-41.0.2251.0_p1 and http://tryit.jssip.net . -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] [BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)

2014-12-22 8:43 GMT+02:00 Juha Heinanen : > Andrey Utkin writes: > >> jssip -> android: no sound from phone to the browser, i see that jssip >> sends "sendonly" attribute for audio in INVITE's SDP. > > same here. perhaps it is better to wait a year or t

[SR-Users] [BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)

f you have configs to make the needed things work, or if you have experience of such things working stable, and can configure it quickly. -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-r

Re: [SR-Users] webrtc clients support using rtpengine

ieger-od/ec6c21c6f9dde1400578 Also, this deploy of http://vmwrtc.intersog.com/sipml5/call.htm works ok, and official one http://sipml5.org/call.htm?svn=222 has more issues. -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 20:38 GMT+02:00 Andrey Utkin : > This works: call from sipml to linphone android: > rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58 > kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46 > ngrep: https://gist.github.com/krieger-od/cb5829be

Re: [SR-Users] webrtc clients support using rtpengine

krieger-od/d677864fcab8c508adde -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 20:09 GMT+02:00 Andrey Utkin : > 2014-12-18 20:05 GMT+02:00 Richard Fuchs : >> Amazon NAT is exactly why I've mentioned it, because on an Amazon >> system, if you don't use the --interface option correctly >> ($INT_IP!$EXT_IP notation), you get exac

Re: [SR-Users] webrtc clients support using rtpengine

ank you, will try with such notation on amazon host now. > But these configs and this CLI line don't match the logs you posted > earlier. Is it also write errors you're getting with those? Will provide you shortly with logs from DigitalOcean VPS instance, configs for which i

Re: [SR-Users] webrtc clients support using rtpengine

rieger/rtpengine/daemon/rtpengine --interface=188.226.241.236 --listen-ng=127.0.0.1:2 -m 3 -M 35000 --foreground --log-stderr -- Andrey Utkin ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org ht

[SR-Users] webrtc clients support using rtpengine

(doesn't work). I would really love to get some quick help from anyone. For direct manual fixing, I can give a couple of hundreds of bucks. Looking forward impatiently for reply from anyone having something to say. -- Andrey Utkin ___ SIP Express R