[asterisk-bugs] [JIRA] (ASTERISK-20650) Asterisk resets ptime value in 200 OK response

Mark Michelson (JIRA) noreply at issues.asterisk.org
Wed Feb 6 11:40:59 CST 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-20650?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=202647#comment-202647 ] 

Mark Michelson commented on ASTERISK-20650:
-------------------------------------------

Yes.

I have traced the problem down to local RTP bridging. When local RTP bridging is used, the smoother that is used to make packetization work properly is bypassed. Instead, the RTP packets are sent directly from the read side to the write side.

I have a quick workaround you can use until the problem is fixed with a patch. For now, you can change your usage of Dial to have an option that prevents local RTP bridging from occurring. You can either set T, t, H, h, X, x, W, w, K, or k on the Dial statements and that will prevent local RTP bridging from occurring.

The problem I have right now is that I don't know if it's possible to change local RTP bridging to use the smoother or if that will completely break local RTP bridging. The other possibility is not to allow local RTP bridging if packetization of the two streams is different.
                
> Asterisk resets ptime value in 200 OK response
> ----------------------------------------------
>
>                 Key: ASTERISK-20650
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20650
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Resources/res_fax
>    Affects Versions: 1.8.17.0
>            Reporter: Maciej Krajewski
>            Severity: Critical
>         Attachments: ast_config.zip, Asterisk_1_4_26_RTP_packetization.pcap, debug_5-autoframing_yes.cap.gz, debug_5-autoframing_yes.log, tmptk_20121102_1527_g02_editcap.cap, tmptk_20121105_1155_g03_edit_filtr.cap
>
>
> Today I have found a problem concerning asterisk 1.8.X
> I've migrated one of my servers from old version of asterisk 1.4 branch to the newest 1.8.
> Everything works fine, but faxes has stopped working.
> I've made some research and what I have found is that asterisk 1.8 has some problems with transcoding ptime, it simply does not do this, Astersik 1.4 was coping with this.
> I've attached the cap file where one of my legs is negotiating connection with ptime 10, the other one with ptime 20.
> However if you look at the payload of RTP packets it is 10 ms every where.

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