[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.

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23425?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217089#comment-217089 ] 

Rusty Newton commented on ASTERISK-23425:
-----------------------------------------

Closing this out since things appear to be working now.

{quote}
Other browsers: Chrome canary (36.0.1923.2 latest), FireFox (28 latest), ie10 with wertc4all not work, i receive this error:

[Apr 1 22:54:53] WARNING[16443][C-00000000]: chan_sip.c:10512 process_sdp: Rejecting secure audio stream without encryption details: audio 51345 RTP/SAVPF 111 103 104 0 8 106 105 13 126
{quote}

For the above issue, you may file a new issue if you like. However as Max said, there will be more changes coming, WebRTC is a moving target and the community of developers is doing their best to keep up with things as time allows.

As always, patches are accepted, you are welcome to contribute with code where you can! :)  Thanks for the report!

> 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: 11.4.0, 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 Ushakov
>            Assignee: Kirill Ushakov
>              Labels: asterisk, sdp
>         Attachments: asterisk_output.txt, chrome_output.txt, extensions.ael.txt, jira-jssip.txt, jira-sipdebug.txt, jira-sipml5.txt, os_kernel_asterisk_chrome.conf, rtp.conf.txt, sip.conf.txt, sipconf.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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