[asterisk-bugs] [JIRA] (ASTERISK-20835) RTP not modified when UAS responds with an OK(200) with other ptime then 20ms

Paolo Compagnini (JIRA) noreply at issues.asterisk.org
Sun Dec 23 10:39:45 CST 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-20835?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=200957#comment-200957 ] 

Paolo Compagnini commented on ASTERISK-20835:
---------------------------------------------

the debug log is a bit long. It contains the asterisk start and the test call. Nothing else.

the pcap has all the related traffic.
All magic happened on the same machine.

"A" is a softphone (sflpohne). running sip on 35060
"B" is SIPp. I used it to reproduce the Szenario. It first happend with an AVM Fritzbox. 'cause It uses always ptime 30. here running on port 15060
Asterisk itselft is running on port 25060

you' ll also see all 4 streams in the pcap
A to asterisk
asterisk to A
asterisk to B
B to asterisk








                
> RTP not modified when UAS responds with an OK(200) with other ptime then 20ms
> -----------------------------------------------------------------------------
>
>                 Key: ASTERISK-20835
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20835
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 10.11.0
>            Reporter: Paolo Compagnini
>            Assignee: Paolo Compagnini
>            Severity: Minor
>         Attachments: debug, extensions.conf, sip_and_rtp.pcap, sip.conf
>
>
> my Asterisk 10 svn recievs an INVITE from A with ptime:20 and invites B also with ptime:20.
> now B sends an OK with ptime:30. The OK from Asterisk to A contains ptime:20.
> But now the RTP stream from asterisk to A is the same as from B to asterisk. That means asterisk doesn't change the stream ( 20 40 20 40 20 40 )and is transmitting with a ptime of 30ms (30 30 30 30 30 ).
> regarding the RFC this shouldn't be an issiue since RFC4566 says:"
>  It should not be necessary to know ptime to decode RTP
>          or vat audio, and it is intended as a recommendation for the
>          encoding/packetisation of audio."
> but this can lead to strange behaviour of A like synchronisations problems.

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators
For more information on JIRA, see: http://www.atlassian.com/software/jira



More information about the asterisk-bugs mailing list