Re: [SR-Users] Transport issue thought

You are right Alex, Linux kernel didn't support SO_REUSEPORT, which allows a socket to be used as source for a tcp connection while is already bound, until version 3.9. Kamailio's parameter tcp_reuse_port, if enabled and your OS has support for SO_REUSEPORT (so not only Linux but FreeBSD, OSX and

Re: [SR-Users] rtpproxy not forwarding rtp packets.

Thanks for replying. Callee is actually a sip provider. I see no firewall or iptables rules on server to prevent rtp following towards callee. Is there anything i can do to see which thing is blocking rtps. Best On Fri, 18 May 2018 at 3:59 AM, Mack Hendricks wrote: > Is

Re: [SR-Users] rtpproxy not forwarding rtp packets.

Is there something blocking RTP traffic from reaching the callee? Is the callee a carrier or the actually endpoint endpoint (aka SIP Phone)? Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com support: +888-907-2085 dSIPRouter

[SR-Users] rtpproxy not forwarding rtp packets.

Greetings list, I have kamailio server behind nat with rptproxy. But i am getting no voice on the call. After taking trace i could see that rtpproxy was getting rtp packets but not packets was being forwarded towards callee side. Though i see in rtpproxy logs, packets being relayed from caller

Re: [SR-Users] Transport issue thought

When an outgoing TCP connection is opened, either a port can be explicitly bound, or it is auto-assigned by the OS's networking stack. I believe Kamailio does the latter and does not offer options to constrain the range. If it does, I'm not aware of any apart from this one:

[SR-Users] Transport issue thought

It appears that the bottom line of my TCP transport not working is that Kamailio is randomly assigning large port numbers to send the TCP traffic out on. I am not able to randomly open high ports for this purpose. Is there a way to tell Kamailio to only use specific ports for this. I have

Re: [SR-Users] Further question on transport issue.

Code snippet #!ifdef WITH_TCP #$ru = $ru + ";transport=tcp"; #force_send_socket(xx.xx.xx.xx:5060); where xx.xx.xx.xx is Kamailio IP-Address xlog("L_INFO","=>Transport TCP"); xlog("L_INFO"," =>du:$du"); xlog("L_INFO"," =>ru:$ru"); if

Re: [SR-Users] Further question on transport issue.

Code snippet #!ifdef WITH_TCP #$ru = $ru + ";transport=tcp"; #force_send_socket(xx.xx.xx.xx:5060); where xx.xx.xx.xx is Kamailio IP-Address xlog("L_INFO","=>Transport TCP"); xlog("L_INFO"," =>du:$du"); xlog("L_INFO"," =>ru:$ru"); if

Re: [SR-Users] cfg script 101

Awesome. Thanks KD On Thursday, May 17, 2018, 11:38:14 AM EDT, Ovidiu Sas wrote: 1. Here's the documentation for while loop: http://www.kamailio.org/wiki/cookbooks/devel/core#while 2. Here are some string helper functions:

[SR-Users] Kamailio sends SIP messages to old websocket after reconnection during call

Hi, We use kamailio as a backend for JSSIP based WebRTC UAs. We use SIP over Weboscket as transport and everythings work fine generally. Some websocket connections to kamailio drop during an active call and JSSIP reconnects immediately. New websocket connection has a different port than the

Re: [SR-Users] cfg script 101

1. Here's the documentation for while loop: http://www.kamailio.org/wiki/cookbooks/devel/core#while 2. Here are some string helper functions: http://www.kamailio.org/wiki/cookbooks/devel/transformations#ssubstr_offset_length

Re: [SR-Users] NatHelper's Keep-alive timeout behind proxy

On 17/05/18 14:23, Daniel Tryba wrote: I noticed the same problem (most important with multiple registrars I didn't like paritioning logic) and my solution was to send my own OPTIONS. We have it running with multiple (3) registrars that replicate with dmq usrloc, works perfectly so far for

Re: [SR-Users] Further question on transport issue.

What is the full and exact RURI of the request immediately prior to relay, and also the value of $du? Can you xlog() both $du and $ru prior to t_relay() and tell us what those values are? Thanks, -- Alex On Thu, May 17, 2018 at 01:36:25PM +, Wilkins, Steve wrote: > Hello, > > I am still

Re: [SR-Users] Further question on transport issue.

Hey Steve, Can provide us a snippet of code on how you are populating the $ru and making invoking the relay? Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com support: +888-907-2085 dSIPRouter - GUI focused on

[SR-Users] Further question on transport issue.

Hello, I am still struggling with getting TCP relay to work. Kamailio is now trying to send TCP but it is using the incorrect port. Kamailio is relaying to Asterisk on xx.xx.xx.xx:some-large-port but Asterisk is listening on xx.xx.xx.xx:5060. I'm not sure why this is happening. I would

Re: [SR-Users] NatHelper's Keep-alive timeout behind proxy

On Thu, May 17, 2018 at 11:43:36AM +0100, Asgaroth wrote: [OPTIONS and seperate proxy/registrar] > Does anyone have any suggestions or pointers on how we can achieve this with > the registrars being seperate from the edge proxies. > > Any pointrs/tips etc would be greatly appreciated. I noticed

Re: [SR-Users] Updated NAT traversal article

You are very welcome. Please let me know if you have any ideas on how it might be usefully expanded. Some others have made good suggestions, but unfortunately they present scope challenges because they lie outside the sphere of server-side NAT traversal as such. On Thu, May 17, 2018 at 11:52:08AM

Re: [SR-Users] Updated NAT traversal article

Alex, Thank you for this blog post, great work, very helpfull information! Much appreciated. On 11/05/18 15:13, Alex Balashov wrote: Hi, I have updated this article with some other topics: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-definitive-guide/ "The

[SR-Users] NatHelper's Keep-alive timeout behind proxy

Hi All, We are looking to enable nathelpers keepalive timeout parameter where it will remove an aor's contact if it doesn't receive a reply from the ua. The problem we have is that our registrar's are seperated from our proxies, so this means nathelper is sending options messages via the