Open GoogleCodeExporter opened 9 years ago
i''d like to add few more comments on this...
in the above scenario, my asterisk server is an amazon web services(cloud)..
i tried the same setup on a local virtual machine and everything is working properly, i can see both ways RTP flowing between chrome and sip..
still same setup is not working in AWS...
can someone please help me with this..
thank you in advance.
Original comment by shad.a.s...@gmail.com
on 4 Mar 2014 at 9:01
I'm running asterisk on EC2 and to make it working on both way audio, I
manually patched asterisk's src/channels/chan_sip.c file to send the IP address
of my EC2 machine.
It is in function: add_ice_to_sdp
Original Line:
ast_str_append(a_buf, 0, "%s ",
ast_sockaddr_stringify_host(&candidate->address));
New line:
ast_str_append(a_buf, 0, "%s ", "[ec2-server-ip-address]");
That's the only way I found it to be working.
Original comment by chetan.g...@gmail.com
on 22 May 2014 at 11:43
Hi,
I'm facing the same situation, then my asterisk server is behind nat (required
ports are forwarded). My Sipml5 + Chromium 34 has sound, but I can't hear
anything on my Cisco phone. rtp debug on asterisk shows bidirectional rtp
stream, but if it try to analize packets with wireshark i can see normal rtp
packets flowing from asterisk to Chromium, and only UDP STUN request packets
flowing from Chromium to asterisk. This mens that ICE session is not
established and don't know how to fix it...
Original comment by virmanta...@lamoda.ru
on 18 Jul 2014 at 7:49
hi,
anyone found solution? please share
Original comment by 2haf...@gmail.com
on 1 Feb 2015 at 5:34
Original issue reported on code.google.com by
shad.a.s...@gmail.com
on 26 Feb 2014 at 7:46Attachments: