[asterisk-bugs] [JIRA] (ASTERISK-19598) Garbled audio using Page app and MulticastRTP channel

Vitaliy Aleksandrov (JIRA) noreply at issues.asterisk.org
Wed Nov 7 04:11:21 CST 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-19598?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199315#comment-199315 ] 

Vitaliy Aleksandrov edited comment on ASTERISK-19598 at 11/7/12 4:10 AM:
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The problem is still present. I have tested MulticastRTP channel with Dial command and it works really great. But when i'm trying to use it with app_page (that uses confbridge) i'm getting a very garbled audio.
Confbridge with SIP channels works great too.

Is there any way to send multicast stream to more that one interface without app_page ?

p.s. all tests were made at asterisk-11
                
      was (Author: vitalik):
    The problem is still present. I have tested MulticastRTP channel with Dial command and it works really great. But when i'm trying to use it with app_page (that uses confbridge) i'm getting a very garbled audio.
Confbridge with SIP channels works great too.

Is there any way to send multicast stream to more that one interface without app_page ?

p.s. all tests are made at asterisk-11
                  
> Garbled audio using Page app and MulticastRTP channel
> -----------------------------------------------------
>
>                 Key: ASTERISK-19598
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-19598
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_page, Channels/chan_multicast_rtp
>    Affects Versions: SVN
>            Reporter: Remi Quezada
>            Severity: Minor
>         Attachments: rtp-page-capture.txt, rtppage.cpt
>
>
> Getting garbled audio with Multicast RTP and Page application.  Multicast RTP works fine with Dial application.  
> I am using the following phones for multicast rtp, all have the same garble audio:
> Cisco spa504G
> Cisco spa303
> SNOM 821
> SIP/256-eng is a Polycom Soundpoint 331
> Able to reproduce with dialplan listed below. I also attached cli debug and an IP capture.  

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