[asterisk-bugs] [JIRA] (ASTERISK-23425) No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.

[Kirill (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23425?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=216199#comment-216199 ] 

Kirill edited comment on ASTERISK-23425 at 3/11/14 1:26 AM:
------------------------------------------------------------

Hello Rusty.
As you say, i moved user "4343" from user.conf to sip.conf
I'm carefully read instruction, checked configuration and didn't found something new.
I'm already use four asterisk servers with sipml5 but they have real IP address and all works fine. This is first server behind the NAT.
Latest (410106) snv build of "Asterisk11" doesn't help me.

Using standart build options: loadable_modules, build_native. 
I seen this issue before and tryed with "don't_optimize" and "better_backtraces" too.
Using latest build sipml5 and trying with demo. 
Nothing helped me.


was (Author: kirus):
Hello Rusty.
As you say, i moved user "4343" from user.conf to sip.conf
I'm carefully read instruction, checked configuration and didn't found something new.
I'm already use four asterisk servers with sipml5 but they have real IP address and all works fine. This is first server behind the NAT.
Latest (410106) snv build of "Asterisk11" doesn't help me.

Using standart build options: loadable_modules, build_native. 
I seen this issue before and tryed with "don't_optimize" and "better_backtraces" too.
Using latest build sipml5 and trying with demo. Nothing helped me.

> No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.
> --------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23425
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23425
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP, Channels/chan_sip/WebSocket, Resources/res_http_websocket, Resources/res_srtp
>    Affects Versions: 11.7.0, 11.8.0
>         Environment: ********** OS ***********
> Distributor ID: Ubuntu
> Description:    Ubuntu 12.04.3 LTS
> Release:        12.04
> Codename:       precise
> **********kernel*************
> Linux version 3.8.0-29-generic (buildd at panlong) (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu5) ) #42~precise1-Ubuntu SMP Wed Aug 14 16:19:23 UTC 2013  x86_64 GNU/Linux
> **********Chrome************
> Chrome 33.0.1750.146 m
> **********Asterisk************
> Astersik version 11.8.0 with SRTP ( ./configure CFLAGS=-fPIC --prefix=/usr ) configured.
> ***Asterisk config users.conf*** 
> [4343]
> canreinvite = no
> type = peer
> host = dynamic
> context = mycontext
> hassip = yes
> hasiax = no
> nat = force_rport,comedia
> qualify = no
> encryption = yes
> avpf = yes
> ;savpf = yes
> language = ru
> videosupport = no
> directmedia = no
> disallow = all
> allow = alaw
> secret = mysecret
> transport = ws,udp
> icesupport = yes
> ***called extention in extensions.ael***
> 7 => {
>      Answer();
>      Playback(hello-world);
>      MusicOnHold(default,300);
>      Hangup();
>      }
>            Reporter: Kirill
>            Assignee: Kirill
>         Attachments: asterisk_output.txt, chrome_output.txt, os_kernel_asterisk_chrome.conf
>
>
> No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.
> In attachments i change my "External IP" and "Domain" to "195.195.195.195"  and "my.domain.com"
> for reproduce the problem, your server must be behind the nat



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