Re: [SR-Users] [sr-dev] Branch 5.0 created

var->offset) != NULL) > ^ > core/cfg/cfg_ctx.c:1707:6: warning: dereferencing type-punned pointer will > break strict-aliasing rules [-Wstrict-aliasing] > replaced[num] = *(char **)(group_inst->vars + var->offset); > ^ > core/cfg/cfg_ctx.c:1713:3: warning

[SR-Users] Kamctl stats

Hello list, Where can I found any information to completely understand what do values returned by 'kamctl stats' represent? Cheers, Alex ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org

Re: [SR-Users] Possible memory leak in mysql driver

ct the output of 'bt full' with gdb and > send it over here. > > Cheers, > Daniel > > > On 15/11/16 22:35, Alexandru Covalschi wrote: >> Hello list, >> >> We’re using dev version of Kamailio: >> version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5

[SR-Users] Possible memory leak in mysql driver

Hello list, We’re using dev version of Kamailio: version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY,

Re: [SR-Users] How to understad that user is de-registrating

Thanks for the info guys, I've fixed my config to suit the correct logic. Thanks again! 2016-06-23 8:40 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>: > Hello, > > On 19/06/16 19:41, Alexandru Covalschi wrote: > > Hello list, > > I need to send to an ex

Re: [SR-Users] WS to WS calls - No ACK received for 200OK

The problem may be with record_route header. Did you set *advertised_address?* 2016-06-21 12:59 GMT+03:00 Amit Patkar : > Hi > > I am using Kamailio as Websocket proxy. > > User 1 & User 2 are registered on Kamailio over WebSocket. > When User 1 calls User 2, User 2 gets ring and

[SR-Users] How to understad that user is de-registrating

and sends REGISTER he is de-register. (Please correct me if I'm wrong.) How can I catch that? Can I use event_route[usrloc:contact-expired]? Thanks in advance! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator tel: +37367398493 web: http://abriss.solutions/ <http://

Re: [SR-Users] Kamailio Problem carrying media

r.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator tel: +37367398493 web: http://abriss.solutions/ <http://abs-telecom.com/> ___ SIP Express Router (SE

Re: [SR-Users] Media server can't return ACK correctly

Thanks everyone 2016-02-25 19:41 GMT+02:00 Alexandru Covalschi <568...@gmail.com>: > Ok guys. The issue was in my misunderstanding of RFC and > advertised_address variable. > Removing advertised_address solved the issue. > > 2016-02-25 17:49 GMT+02:00 Alberto Sagredo <a

Re: [SR-Users] Media server can't return ACK correctly

gt; > Record-Route: <sip:PUBLIC_IP;r2=on;lr=on;ftag=as2c0c55b9> > Record-Route: <sip:PRIVATE_IP;r2=on;lr=on;ftag=as2c0c55b9> > > And ACKS will go to right place.. > > > 2016-02-25 16:43 GMT+01:00 Alexandru Covalschi <568...@gmail.com>: > >> force_send_socket is a

Re: [SR-Users] Media server can't return ACK correctly

No other rr params defined so double rr is default - enabled. What do you mean by "force traffic" - how to do that? Every other request (excep BYE - same problem with it) flows OK. 2016-02-25 11:49 GMT+02:00 Alexandru Covalschi <568...@gmail.com>: > Hi, thanks for answer >

Re: [SR-Users] Media server can't return ACK correctly

guration about Public IP and Private IP? > > Do you use advertise? > > Maybe you need to force Outbound traffic to Public IP Socket and inside > traffic to Private IP . > > Do you have double record routing? > > BR > > 2016-02-25 1:24 GMT+01:00 Alexandru Covalschi

[SR-Users] Media server can't return ACK correctly

pwSigF I'm seeking help with that - what parameter I need to change/add to solve that? Maybe it's a networking problem - but why then ACK reaches Freeswitch and all other requests flow OK? Thanks in advance, Alex -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator

Re: [SR-Users] Kamailio and NAT

the source port. > > Contact: <sip:9@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1> > > --- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp", "udp") in > order to received new transactions or should I follow a different > procedure??? > > Thank you > > > > > _

[SR-Users] IMC module

"111" from user 1001 and imc_manager catches it - I receive 500 command error. Why? :/ All that is working on top of sipjs-demo. -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-t

Re: [SR-Users] IMC module

sl_send_reply("500", "command error"); exit; } This allows me to receive system messages - but I can't get any messages from clients. 2015-12-15 18:43 GMT+02:00 Alexandru Covalschi <568...@gmail.com>: > Hello again > First of all I wanted to ask if someone ever implem

Re: [SR-Users] IMC module

"chat-3500") > { > if(imc_manager()) > sl_send_reply("200", "ok"); > else > sl_send_reply("500", "command error"); > exit; > } > This allows me to receive system messages - but I can't get any messages

[SR-Users] Relaying failure codes back to initial server

ended the call. Now Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me how to achieve that? Thanks in advance -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com

Re: [SR-Users] Relaying failure codes back to initial server

GA 30346 > United States > > Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > Sent from my BlackBerry. > *From: *Alexandru Covalschi > *Sent: *Tuesday, December 15, 2015 05:03 > *To: *Kamailio (SER)

Re: [SR-Users] Relaying failure codes back to initial server

ns you >> could be taking somewhere to drop it, such as in an onreply_route? >> >> ACKs to negative final replies are hop-by-hop, so the ACK you're seeing >> directly from the proxy to the UAS is normal. >> >> -- >> Alex Balashov | Principal | Evariste S

[SR-Users] Kamailio as UAC

, "Remote asked for authentication"); uac_auth(); } to MANAGE_FAILURE or MANAGE_REPLY route Kamailio can't start. Is that even possible? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/ ___

Re: [SR-Users] Kamailio as UAC

I saw that, but 1. It doesn't work in failure_route (MANAGE_FAILURE from std. config) either 2. My question was more general - is it even possible to do what I need with Kamailio 2015-11-27 13:24 GMT+02:00 Daniel Tryba <d.tr...@pocos.nl>: > On Friday 27 November 2015 13:01:19 Alexandru

Re: [SR-Users] Kamailio as UAC

Well I tried but didn't work for me :( however problem is solved using other voip provider. Thanks for help! 27 нояб. 2015 г. 14:09 пользователь "Daniel Tryba" <d.tr...@pocos.nl> написал: > On Friday 27 November 2015 13:50:36 Alexandru Covalschi wrote: > > I saw that, b

Re: [SR-Users] Grab users password from WWW-Auth header

pt, or maybe I can use some built-it functions? 2015-11-13 19:52 GMT+02:00 Alexandru Covalschi <568...@gmail.com>: > Many thanks for you help Sebastian! > > 2015-11-13 19:13 GMT+02:00 Sebastian Damm <d...@sipgate.de>: > >> >> On Fri, Nov 13, 2015 at 3:43 PM, Alexan

Re: [SR-Users] Authorization selects

UPD: proxy_auth doesn't work either, however I'm sure I have WWW-Auth, not Proxy-Auth :) 2015-11-16 16:23 GMT+02:00 Alexandru Covalschi <568...@gmail.com>: > Hello everyone! > > I need to extract values from authentication header, but > 408. @authorization["string"] &

[SR-Users] Authorization selects

QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id

Re: [SR-Users] Grab users password from WWW-Auth header

Many thanks for you help Sebastian! 2015-11-13 19:13 GMT+02:00 Sebastian Damm <d...@sipgate.de>: > > On Fri, Nov 13, 2015 at 3:43 PM, Alexandru Covalschi <568...@gmail.com> > wrote: > >> What if I don't need a plaintext password on Kamailio? I mean, I don't >&

Re: [SR-Users] Grab users password from WWW-Auth header

So it should be like ... if (!has_credentials("myrealm")) { www_challenge("$td", "1"); } else { if (!my_script()){ sl_send_reply("401", "Not Authorized"); } } ... 2015-11-13 16:13 GMT+02:00 Alexandru Coval

Re: [SR-Users] Grab users password from WWW-Auth header

password. > > Best Regards, > Sebastian > > > > On Fri, Nov 13, 2015 at 3:13 PM, Alexandru Covalschi <568...@gmail.com> > wrote: > >> simple send_reply("200", "OK");, sorry >> >> 2015-11-13 16:02 GMT+02:00 Alexandru Covalschi &l

Re: [SR-Users] Grab users password from WWW-Auth header

unctions in a pseudo variable. > > > http://www.kamailio.net/docs/modules/4.3.x/modules/auth.html#auth.f.pv_www_authenticate > > Best Regards, > Sebastian > > On Fri, Nov 13, 2015 at 2:14 PM, Alexandru Covalschi <568...@gmail.com> > wrote: > >> UPD: If

Re: [SR-Users] Grab users password from WWW-Auth header

simple send_reply("200", "OK");, sorry 2015-11-13 16:02 GMT+02:00 Alexandru Covalschi <568...@gmail.com>: > Thanks for your reply! But the problem is - I need to provide to API > user's login and password. Kamailio doesn't know them. So my idea was to > tra

[SR-Users] Grab users password from WWW-Auth header

;); } The main problem is - how can I grab or compare users password? I know nonce, which I understand is MD5 salt. Can I, for example, grab users password from API, then grab the MD5 string and the nonce user sent me, calculate MD5 on base of API password and nonce - and then compare MD5 strings

Re: [SR-Users] Grab users password from WWW-Auth header

UPD: If upper method is possible - I assume I can check if message has Auth header using if (has_credentials("myrealm")) { ... } Can you please specify how to grab it? 2015-11-13 15:08 GMT+02:00 Alexandru Covalschi <568...@gmail.com>: > Hello! > My problem

[SR-Users] Kamailio + Rtpengine < - > Freeswitch + ZRTP

uality is sometimes bad and the delay is too big. Anyone has any ideas? Kamailio 4.3, FS 1.4 <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and sy

Re: [SR-Users] authorize a client in bulk registration

ed. > > Your assistance in this matter is greatly appreciated > > Thanks, > Al > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.o

Re: [SR-Users] Using PSTN as fallback

@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/ ___ SIP Express Router (SER) and Kamailio

Re: [SR-Users] unsupported route_type 64

in branch 4.1 (part of 4.1.5). Thanks. ___ 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 -- Alexandru Covalschi ABRISS

[SR-Users] Kamailio TLS configuration

= no Make everything work. Cross-domain calling is essential and I'm just trying to figure out - what's the problem? Is that my certificate, is that ostel.co certificate or it is just the way it should be? Thanks! -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone

Re: [SR-Users] Kamailio TLS configuration

compiled with gcc 4.9.2 openssl version OpenSSL 1.0.1k 8 Jan 2015 2015-08-28 20:01 GMT+03:00 Alexandru Covalschi 568...@gmail.com: Hello! I'm having problems with Kamailio configuration with TLS. Or, maybe, that's my misunderstanding about how it should work. So, the issue - inbound TLS works

Re: [SR-Users] Kamailio TLS configuration

And server is under Amazon EC2, but that shouldn't really make any sense 2015-08-29 0:11 GMT+03:00 Alexandru Covalschi 568...@gmail.com: Forgot to add cat /etc/issue Debian GNU/Linux 8 \n \l kamailio -V version: kamailio 4.3.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP

Re: [SR-Users] Help with sip balancer

___ 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 -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http

Re: [SR-Users] Help with sip balancer

Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already included. Also you can use LCR for routing calls to different providers, a simple guide can be found here http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ 2015-08-12 0:41 GMT+03:00 Alexandru Covalschi 568

Re: [SR-Users] Debian/jessie and 4.3.1 default config doesn't work

Hmf... I saw the advice to put them on /tmp/ somewhere on mailing lists and had same thoughts. Thanks, will fix that on my servers! 2015-08-10 14:16 GMT+03:00 Daniel Tryba d.tr...@pocos.nl: On Monday 10 August 2015 13:12:12 Alexandru Covalschi wrote: Shouldn't they be /tmp/kamailio_fifo

Re: [SR-Users] Debian/jessie and 4.3.1 default config doesn't work

) ___ 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 -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com

Re: [SR-Users] LCR Routing - Cost based routing

module. DanB ___ 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 -- Alexandru Covalschi ABRISS-Solutions VoIP engineer

Re: [SR-Users] LCR Routing - Cost based routing

/listinfo/sr-users -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http

Re: [SR-Users] LCR Routing - Cost based routing

Or, well, see that guide http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we have priority and weight on LCR module 2015-08-09 12:37 GMT+03:00 Alexandru Covalschi 568...@gmail.com: try using CGRateS 2015-08-09 12:06 GMT+03:00 Arun Kumar mi2a...@gmail.com: Hi

Re: [SR-Users] LCR Routing - Cost based routing

however you can try building LCR based on prefix and weight, why not? 2015-08-09 18:52 GMT+03:00 Alexandru Covalschi 568...@gmail.com: I know CGRateS allow cost-based LCR for Kamailio. Maybe there are some internal Kamailio modules, but I don't know about them 2015-08-09 17:10 GMT+03:00 Arun

Re: [SR-Users] LCR Routing - Cost based routing

, Aug 9, 2015 at 3:09 PM, Alexandru Covalschi 568...@gmail.com wrote: Or, well, see that guide http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we have priority and weight on LCR module 2015-08-09 12:37 GMT+03:00 Alexandru Covalschi 568...@gmail.com: try using CGRateS 2015-08

Re: [SR-Users] LCR Routing - Cost based routing

? is there any avaliable way to automate or we have to rewrite/modify the lcr/drouting module for rate selection , On Sun, Aug 9, 2015 at 9:22 PM, Alexandru Covalschi 568...@gmail.com wrote: however you can try building LCR based on prefix and weight, why not? 2015-08-09 18:52 GMT+03:00

Re: [SR-Users] Rtpengine/rtpproxy and zrtp

thanks! 2015-08-06 22:29 GMT+03:00 Frank Carmickle fr...@carmickle.com: Zrtp passes through rtpengine just fine. --FC Sent from my 6 plus On Aug 6, 2015, at 14:12, Alexandru Covalschi 568...@gmail.com wrote: Sorry if writing to wrong mailing list, I am very limited to traffic now amd

[SR-Users] Rtpengine/rtpproxy and zrtp

Sorry if writing to wrong mailing list, I am very limited to traffic now amd don't know if there is any for rtpproxy/rtpengine. My question is - can they support ZRTP at least in pass-through mode? Will rtpengine fail on trying to recognize unknown SDP fields?

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

I got bridging working well on internal interfaces in case of simple SIP calls on a bit other configuration. But editing this config to support WebRTC causes same problems. I need internal interfaces on asterisk to completely close external ones (Security etc.).

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

{ xlog(L_NOTICE,= $fU has NO WEBSOCKETS); rtpengine_manage(replace-origin replace-session-connection RTP/AVP); } } 2015-06-24 16:14 GMT+03:00 Alexandru Covalschi 568...@gmail.com: Heh... Well, I still have troubles with my

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

Asterisk localip=10.0.0.87, sorry 2015-06-24 16:24 GMT+03:00 Alexandru Covalschi 568...@gmail.com: Ok, so my scheme. Kamailio and Asterisk are in Amazon EC2 Kamailio externip=54.197.230.121 localip=10.145.45.103 Asterisk localip=10.145.45.103, externip doesn't matter Call should flow like

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

what IP and what you would expect there? It is not easy to digests lots of logs and also guess what would you expect to happen... Cheers, Daniel On 24/06/15 15:14, Alexandru Covalschi wrote: Heh... Well, I still have troubles with my configuration. And in SDP media adress is Amazon

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568...@gmail.com: Well.. Guys, sorry, it was totally my fault. I just used VPN. 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com: I used https://github.com/caruizdiaz/kamailio-ws

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

without fix_nated_contact error behaviour is the same maybe I should upgrade to 4.3 ? 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com: Here's the trace on port which I use for ws server. Don't look at fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

. Can you look at javascript debug console in the browser to see what is printing? Daniel On 23/06/15 17:23, Alexandru Covalschi wrote: without fix_nated_contact error behaviour is the same maybe I should upgrade to 4.3 ? 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

GMT+03:00 Alexandru Covalschi 568...@gmail.com: Here is it http://pastebin.com/JkkM4M5m 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com: There are no major changes in 4.3 comparing with 4.2 in regards to websocket -- the implementation is quite mature for a long time

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

I solved the SIP voice trouble, but WebRTC problem still exists. What kind of trace I must do to make my post more informative? 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com: Hello, On 23/06/15 04:10, Alexandru Covalschi wrote: Hello. I'm trying to set up this (v

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

Here's the trace on port which I use for ws server. Don't look at fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

Well.. Guys, sorry, it was totally my fault. I just used VPN. 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com: I used https://github.com/caruizdiaz/kamailio-ws configuration that 100% works on other then Amazon EC2 environment and I still get this error. Maybe it is somehow

[SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

(trust-address replace-origin replace-session-connection ICE=force RTP/AVP); Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com

Re: [SR-Users] SIP-over-Websocket Load Balancing

thanks, will try that 2015-06-15 14:07 GMT+03:00 Juha Heinanen j...@tutpro.com: Alexandru Covalschi writes: sorry, i thought you use registrar/usrloc modules Well, I do use them - so if you could explain in which table does Kamailio write the user's proto and which flags I can use

[SR-Users] Kamailio license usage

Maybe it may be an offtopic, but I'm not really into legal issues - so I'm sorry if this message is not fully related to this mailing list. Can I use Kamailio to provide VoIP backend for kind of CRM system in case of SaaS? --- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system

Re: [SR-Users] Kamailio license usage

. Cheers, Daniel On 15/06/15 18:09, Alexandru Covalschi wrote: Maybe it may be an offtopic, but I'm not really into legal issues - so I'm sorry if this message is not fully related to this mailing list. Can I use Kamailio to provide VoIP backend for kind of CRM system in case of SaaS

Re: [SR-Users] SIP-over-Websocket Load Balancing

replace-origin replace-session-connection force trust-address RTP/AVP); } } 2015-06-14 22:24 GMT+03:00 Juha Heinanen j...@tutpro.com: Alexandru Covalschi writes: you don't need a database for that. you can use location table flags Can you please describe how to do that? I

Re: [SR-Users] SIP-over-Websocket Load Balancing

to kamailiio act proxy server Any idea how i can achieve thid On Sun, Jun 14, 2015 at 12:24 AM, Alexandru Covalschi 568...@gmail.com wrote: Well, I performed that by creating a media relay consisting of 2 freeswitches and using rtpengine. You just need to handle WebRTC by kamailio using

Re: [SR-Users] Websocket Load Balancing

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 -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com

Re: [SR-Users] SIP-over-Websocket Load Balancing

___ 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 -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs

Re: [SR-Users] SIP-over-Websocket Load Balancing

Sorry, a mistake: on outgoing webrtc it MUST have RTP/SAVP or RTP/SAVPF 2015-06-13 21:54 GMT+03:00 Alexandru Covalschi 568...@gmail.com: Well, I performed that by creating a media relay consisting of 2 freeswitches and using rtpengine. You just need to handle WebRTC by kamailio using