[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:all-tabpanel ]

Kirill updated ASTERISK-23425:
------------------------------

    Comment: was deleted

(was: in sip.conf.txt
udp port changed
domain name changed to "my.domain.com"
)

> 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, 11.8.1
>         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, extensions.ael.txt, os_kernel_asterisk_chrome.conf, rtp.conf.txt, sip.conf.txt
>
>
> 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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