[asterisk-bugs] [JIRA] (ASTERISK-21411) Far end Re-invites Asterisk without SDP (ACKs with SDP) - Asterisk does not modify port of RTP stream - Ignores ACK due to no SDP version change

[Michael L. Young (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21411?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205497#comment-205497 ] 

Michael L. Young commented on ASTERISK-21411:
---------------------------------------------

I agree with Rusty's assessment above.  There is one comment that needs to be clarified.

{quote}
The ContactRoute server doesn't change the SDP version in it's ACK from the previous SDP it sent. Asterisk notes in the debug after the first ACK with SDP back from the ContactRoute server:
{quote}

Actually, it does change it but it changes it incorrectly.
The ACK changes the SDP "o=" from:
{quote}
o=root 2146949866 2146949869 IN IP4 172.27.172.3
{quote}

To:
{quote}
o=default 28 2 IN IP4 172.27.172.121
{quote}

First, the "o=" header is changed incorrectly according the RFC.

Second, Asterisk is looking at the session version in that line to determine if the SDP needs to be processed and since the session version didn't increment (it went backwards from 2146949869 to 2), Asterisk ignores the SDP.  Therefore, it doesn't change the audio port resulting in one-way audio.

Please try setting ignoresdpversion=yes, like Rusty pointed out, to see if it fixes this for you.  The RTP Engine had changes between 1.4 and 1.8.  Asterisk 1.8 is actually working the correct way.
                
> Far end Re-invites Asterisk without SDP (ACKs with SDP) - Asterisk does not modify port of RTP stream - Ignores ACK due to no SDP version change
> ------------------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-21411
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21411
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.21.0, 11.2.0
>         Environment: CentOS release 5.9 (Final)
>            Reporter: Olivier Lambert
>            Assignee: Olivier Lambert
>         Attachments: issue21411.zip
>
>
> A re-invite without SDP is causing the voice to be established in one way. The exact same test done on asterisk 1.4.38 is not giving the issue, creating a two ways audio as it should. Pcaps of the two tests are available at http://www.misterlambert.net/pcaps.zip. Reinvite is located at frame 5127 for the 1.4 test while it is located at frame 7897 for the 1.8. The one way audio is caused by the missing RTP flow visible at frame 5166.

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