[asterisk-bugs] [JIRA] (ASTERISK-28731) Directmedia Reinvites have SDP with codecs from configuration not negotiation

Christian Berger (JIRA) noreply at issues.asterisk.org
Mon Feb 10 05:42:25 CST 2020


     [ https://issues.asterisk.org/jira/browse/ASTERISK-28731?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Christian Berger updated ASTERISK-28731:
----------------------------------------

    Attachment: clean.pcapng
                ast_c_config.tar.gz
                ast_b_config.tar.gz
                ast_a_config.tar.gz

This shows the Setup.

Asterisk A has the IP 192.168.11.65
Asterisk B has the IP 192.168.11.68
Asterisk C has the IP 192.168.11.64

> Directmedia Reinvites have SDP with codecs from configuration not negotiation
> -----------------------------------------------------------------------------
>
>                 Key: ASTERISK-28731
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28731
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: 13.31.0
>            Reporter: Christian Berger
>         Attachments: ast_a_config.tar.gz, ast_b_config.tar.gz, ast_c_config.tar.gz, clean.pcapng
>
>
> When Asterisk sends out Re-INVITES to 2 connected parties in order to get out of the media stream, the SDP it offers does not reflect the negotiated SDP, but the one configured in sip.conf
> I have written an overview about the issue here:
> https://community.asterisk.org/t/chan-sip-direct-rtp-codec-negotiation-problem/82425
> In short the situation is like this:
> I have made a minimalistic setup in order to examine an unreleated bug. In this setup I have 3 Asterisk instances (A,B and C) running on 3 separate computers.
> Asterisk A is configured to use RTP-Events, Asterisk C is configured to only use Inband DTMF. Asterisk B is configured to accept both and to Re-Invite both parties in order to get out of the media stream when it is possible.
> When a call comes from A to B, B plays a message and forwards it to C. The A-B leg will include RTP-Events, the B-C leg will not include RTP-Events. However shortly after this, Asterisk Re-Invites A and C to get out of the media stream.
> In these Re-Invites it will offer RTP-Events to both sides, making A believe that C can accept RTP-Events when it fact cannot do that.



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