Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP. In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.
you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it.
Cheers, Daniel
In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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/
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
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 miconda@gmail.com:
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.
you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it.
Cheers, Daniel
In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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
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 568691@gmail.com:
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 miconda@gmail.com:
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.
you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it.
Cheers, Daniel
In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
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 568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
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 miconda@gmail.com:
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.
you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it.
Cheers, Daniel
In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.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.
Looks like websocket connection is not available. 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 <568691@gmail.com mailto:568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@gmail.com>>: 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 <miconda@gmail.com <mailto:miconda@gmail.com>>: Hello, On 23/06/15 04:10, Alexandru Covalschi wrote:Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it. Cheers, DanielIn case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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 <tel:%2B37367398493> web: http://abs-telecom.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/-- 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://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@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.
Looks like websocket connection is not available. 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 568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
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 miconda@gmail.com:
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.
you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it.
Cheers, Daniel
In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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
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 related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@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.
Looks like websocket connection is not available. 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 568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
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 miconda@gmail.com :
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.
you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it.
Cheers, Daniel
In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568691@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 related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@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.
Looks like websocket connection is not available. 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 568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
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 <miconda@gmail.com
:
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme
I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.
you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it.
Cheers, Daniel
In case of WebRTC I get lot's of erros:
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
The call reaches Asterisk, but not vice-versa. No media is being transferred.
Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
Heh... Well, I still have troubles with my configuration. And in SDP media adress is Amazon public interface - but rtpengine has replace-origin replace-session-connection session, so it must be local address. Any ideas? Asterisk log http://pastebin.com/MFt9V9qK Kamailio log http://pastebin.com/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz
2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568691@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 related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@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.
Looks like websocket connection is not available. 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 568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
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 < miconda@gmail.com>:
> Hello, > > On 23/06/15 04:10, Alexandru Covalschi wrote: > > Hello. I'm trying to set up this (v 4.2 stable): > peer <--> ec2 <--kamailio+rtpengine--> asterisk > scheme > > I use advertised adress for SIP and WS connections. > The problem is that on SIP I get one way audio - I can receive > audio from asterisk, but I can't transmit audio there - my SIP UA tries to > send data to Kamailio-s local EC2 IP. > > > you should grab a ngrep trace on server to see what happens in the > signaling in order to be able to provide some hints on solving it. > > Cheers, > Daniel > > In case of WebRTC I get lot's of erros: > > Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> > [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for > WebSocket could not be found > Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> > [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via > header > Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> > [forward.c:584]: forward_request(): building failed > Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl > [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm > terribly sorry, server error occurred (1/SL) > > The call reaches Asterisk, but not vice-versa. No media is being > transferred. > > Rtpengine flags I use: > For SIP: rtpengine_manage("trust-adress replace-origin > replace-session-connection RTP/AVP"); > For WS: rtpengine_manage("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/ > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > > > _______________________________________________ > 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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
Can you specify exactly which side received 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 public interface - but rtpengine has replace-origin replace-session-connection session, so it must be local address. Any ideas? Asterisk log http://pastebin.com/MFt9V9qK Kamailio log http://pastebin.com/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz
2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568691@gmail.com mailto:568691@gmail.com>:
Well.. Guys, sorry, it was totally my fault. I just used VPN. 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@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 related to NAT traversal? Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@gmail.com>>: Here is it http://pastebin.com/JkkM4M5m 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@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. Looks like websocket connection is not available. 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 <568691@gmail.com <mailto:568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@gmail.com>>: 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 <miconda@gmail.com <mailto:miconda@gmail.com>>: Hello, On 23/06/15 04:10, Alexandru Covalschi wrote:Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it. Cheers, DanielIn case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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 <tel:%2B37367398493> web: http://abs-telecom.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.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 that: webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip but now it's webrtc --> kamailio-externip --> kamailio--localip --> asterisk-localip --> kamailio-externip --> peer
I have the voice, but it's wrong scheme, and Asterisk drops call because of retransmissions failure
2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla miconda@gmail.com:
Can you specify exactly which side received 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 public interface - but rtpengine has replace-origin replace-session-connection session, so it must be local address. Any ideas? Asterisk log http://pastebin.com/MFt9V9qK Kamailio log http://pastebin.com/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz
2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568691@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 related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@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.
Looks like websocket connection is not available. 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 568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
> 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 < > miconda@gmail.com>: > >> Hello, >> >> On 23/06/15 04:10, Alexandru Covalschi wrote: >> >> Hello. I'm trying to set up this (v 4.2 stable): >> peer <--> ec2 <--kamailio+rtpengine--> asterisk >> scheme >> >> I use advertised adress for SIP and WS connections. >> The problem is that on SIP I get one way audio - I can receive >> audio from asterisk, but I can't transmit audio there - my SIP UA tries to >> send data to Kamailio-s local EC2 IP. >> >> >> you should grab a ngrep trace on server to see what happens in the >> signaling in order to be able to provide some hints on solving it. >> >> Cheers, >> Daniel >> >> In case of WebRTC I get lot's of erros: >> >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> >> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for >> WebSocket could not be found >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> >> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via >> header >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> >> [forward.c:584]: forward_request(): building failed >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl >> [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm >> terribly sorry, server error occurred (1/SL) >> >> The call reaches Asterisk, but not vice-versa. No media is being >> transferred. >> >> Rtpengine flags I use: >> For SIP: rtpengine_manage("trust-adress replace-origin >> replace-session-connection RTP/AVP"); >> For WS: rtpengine_manage("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/ >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.com >> >> >> _______________________________________________ >> 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/ >
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
Asterisk localip=10.0.0.87, sorry
2015-06-24 16:24 GMT+03:00 Alexandru Covalschi 568691@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 that: webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip but now it's webrtc --> kamailio-externip --> kamailio--localip --> asterisk-localip --> kamailio-externip --> peer
I have the voice, but it's wrong scheme, and Asterisk drops call because of retransmissions failure
2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla miconda@gmail.com:
Can you specify exactly which side received 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 public interface - but rtpengine has replace-origin replace-session-connection session, so it must be local address. Any ideas? Asterisk log http://pastebin.com/MFt9V9qK Kamailio log http://pastebin.com/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz
2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568691@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 related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <miconda@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.
Looks like websocket connection is not available. 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 568691@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 > establish a ws connection properly. Client is SIPML5 demo phone > http://pastebin.com/LvAk2HkP > > 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com: > >> 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 < >> miconda@gmail.com>: >> >>> Hello, >>> >>> On 23/06/15 04:10, Alexandru Covalschi wrote: >>> >>> Hello. I'm trying to set up this (v 4.2 stable): >>> peer <--> ec2 <--kamailio+rtpengine--> asterisk >>> scheme >>> >>> I use advertised adress for SIP and WS connections. >>> The problem is that on SIP I get one way audio - I can receive >>> audio from asterisk, but I can't transmit audio there - my SIP UA tries to >>> send data to Kamailio-s local EC2 IP. >>> >>> >>> you should grab a ngrep trace on server to see what happens in >>> the signaling in order to be able to provide some hints on solving it. >>> >>> Cheers, >>> Daniel >>> >>> In case of WebRTC I get lot's of erros: >>> >>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: >>> <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) >>> for WebSocket could not be found >>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> >>> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via >>> header >>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> >>> [forward.c:584]: forward_request(): building failed >>> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl >>> [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm >>> terribly sorry, server error occurred (1/SL) >>> >>> The call reaches Asterisk, but not vice-versa. No media is being >>> transferred. >>> >>> Rtpengine flags I use: >>> For SIP: rtpengine_manage("trust-adress replace-origin >>> replace-session-connection RTP/AVP"); >>> For WS: rtpengine_manage("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/ >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Book: SIP Routing With Kamailio - http://www.asipto.com >>> >>> >>> _______________________________________________ >>> 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/ >> > > > > -- > Alexandru Covalschi > ABRISS-Solutions > VoIP engineer and system administrator > phone: +37367398493 > web: http://abs-telecom.com/ >
-- 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
Do you have proper routing rules between the local ips of kamailio and asterisk? Why aren't you use only external IPs if they are on different servers? Asterisk has also the option to set external ip. It can reduce the complexity of doing bridging of signaling and rtp. Once you get that working you can start adding bridging step by step.
Cheers, Daniel
On 24/06/15 15:25, Alexandru Covalschi wrote:
Asterisk localip=10.0.0.87, sorry
2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568691@gmail.com mailto:568691@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 that: webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip but now it's webrtc --> kamailio-externip --> kamailio--localip --> asterisk-localip --> kamailio-externip --> peer I have the voice, but it's wrong scheme, and Asterisk drops call because of retransmissions failure 2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@gmail.com>>: Can you specify exactly which side received 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 public interface - but rtpengine has replace-origin replace-session-connection session, so it must be local address. Any ideas? Asterisk log http://pastebin.com/MFt9V9qK Kamailio log http://pastebin.com/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@gmail.com>>: Well.. Guys, sorry, it was totally my fault. I just used VPN. 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@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 related to NAT traversal? Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@gmail.com>>: Here is it http://pastebin.com/JkkM4M5m 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <miconda@gmail.com <mailto:miconda@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. Looks like websocket connection is not available. 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 <568691@gmail.com <mailto:568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568691@gmail.com <mailto:568691@gmail.com>>: 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 <miconda@gmail.com <mailto:miconda@gmail.com>>: Hello, On 23/06/15 04:10, Alexandru Covalschi wrote:Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.you should grab a ngrep trace on server to see what happens in the signaling in order to be able to provide some hints on solving it. Cheers, DanielIn case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("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 <tel:%2B37367398493> web: http://abs-telecom.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto: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 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/ -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 <tel:%2B37367398493> web: http://abs-telecom.com/-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
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.).
Also, an interesting thing - if you can see in Kamailio log, a check of the proto of user "300" is being made. But 300 is $tU, and $tU proto is being checked only if source IP is asterisks IP.
Here's the part of config where rtpengine is engaged (in NATmanage route)
if((src_ip==10.0.0.87)) { xlog("L_NOTICE","====== select proto from sipusers where name=$tU"); sql_xquery("ca_asterisk", "select proto from sipusers where name=$tU", "ra"); xlog("L_NOTICE","===== $tU has proto $xavp(ra=>proto)"); if ($xavp(ra=>proto)=="ws") { xlog("L_NOTICE","===== $tU has WEBSOCKETS");
rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); } else { xlog("L_NOTICE","===== $tU has NO fucken WEBSOCKETS"); rtpengine_manage("trust-address replace-origin replace-session-connection"); } } else { xlog("L_NOTICE","====== select proto from sipusers where name=$fU"); sql_xquery("ca_asterisk", "select proto from sipusers where name=$fU", "ra"); if ($xavp(ra=>proto)=="ws") {
xlog("L_NOTICE","===== $fU has WEBSOCKETS"); rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); } else { 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 568691@gmail.com:
Heh... Well, I still have troubles with my configuration. And in SDP media adress is Amazon public interface - but rtpengine has replace-origin replace-session-connection session, so it must be local address. Any ideas? Asterisk log http://pastebin.com/MFt9V9qK Kamailio log http://pastebin.com/jZceP2Rn Javascript log http://pastebin.com/4ZLePyKz
2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568691@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 related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
Here is it http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla miconda@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.
Looks like websocket connection is not available. 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 568691@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 establish a ws connection properly. Client is SIPML5 demo phone http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568691@gmail.com:
> 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 < > miconda@gmail.com>: > >> Hello, >> >> On 23/06/15 04:10, Alexandru Covalschi wrote: >> >> Hello. I'm trying to set up this (v 4.2 stable): >> peer <--> ec2 <--kamailio+rtpengine--> asterisk >> scheme >> >> I use advertised adress for SIP and WS connections. >> The problem is that on SIP I get one way audio - I can receive >> audio from asterisk, but I can't transmit audio there - my SIP UA tries to >> send data to Kamailio-s local EC2 IP. >> >> >> you should grab a ngrep trace on server to see what happens in the >> signaling in order to be able to provide some hints on solving it. >> >> Cheers, >> Daniel >> >> In case of WebRTC I get lot's of erros: >> >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> >> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for >> WebSocket could not be found >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> >> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via >> header >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> >> [forward.c:584]: forward_request(): building failed >> Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl >> [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm >> terribly sorry, server error occurred (1/SL) >> >> The call reaches Asterisk, but not vice-versa. No media is being >> transferred. >> >> Rtpengine flags I use: >> For SIP: rtpengine_manage("trust-adress replace-origin >> replace-session-connection RTP/AVP"); >> For WS: rtpengine_manage("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/ >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.com >> >> >> _______________________________________________ >> 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/ >
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com
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/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/
-
Alexandru Covalschi -
Daniel-Constantin Mierla