[asterisk-bugs] [JIRA] (ASTERISK-27441) PJSIP: Need ignoresdpversion option that chan_sip has for parity.

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27441?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=240314#comment-240314 ] 

Joshua Colp edited comment on ASTERISK-27441 at 11/28/17 10:51 AM:
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[~lvl] You can't modify active media state. Once a media state transitions to active it is treated as immutable and is supposed to be guaranteed to not change (and thus not need protection from other threads modifying it, which removes deadlock potential). You can only create a new media state and replace the active media state.


was (Author: jcolp):
[~lvl] You can't modify active media state. Once a media state transitions to active it is treated as immutable and is supposed to be guaranteed to not change (and thus not need protection from other threads modifying it, which removes deadlock potential).

> PJSIP: Need ignoresdpversion option that chan_sip has for parity.
> -----------------------------------------------------------------
>
>                 Key: ASTERISK-27441
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27441
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 15.1.2
>            Reporter: lvl
>            Assignee: Unassigned
>         Attachments: asterisk-sdpversion-log.txt
>
>
> The scenario:
> 1. Asterisk dials outbound using chan_pjsip
> 2. The remote party sends an 183, with IP address 1.2.3.4 in the SDP *1
> 3. RTP is now flowing between Asterisk and 1.2.3.4
> 4. The remote party sends a 200, with a *new* IP address 5.6.7.8 in the SDP *2
> 5. Asterisk is not applying the new SDP details. 5.6.7.8 will start sending RTP to Asterisk, which is rejected:
> {code}
> [Nov 22 14:29:48] DEBUG[8280][C-00000001]: res_rtp_asterisk.c:5743 ast_rtp_read: 0x7f021c0286e0 -- Received RTP packet from 5.6.7.8:9898, dropping due to strict RTP protection.
> {code}
> The following log lines hint at the root cause:
> {code}
> [Nov 22 14:29:46] DEBUG[8237]: res_pjsip_session.c:3187 handle_incoming: Received response
> [Nov 22 14:29:46] DEBUG[8237]: res_pjsip_session.c:3171 handle_incoming_response: Response is 200 OK
> [Nov 22 14:29:46] DEBUG[8237]: res_pjsip_session.c:798 handle_negotiated_sdp: Pending topology was NULL for channel 'PJSIP/proxy-00000001'
> {code}
> The message is triggered by the code in https://github.com/asterisk/asterisk/blob/15.1/res/res_pjsip_session.c#L796. Because there is no "session->pending_media_state->topology" at this point, the *handle_negotiated_sdp* function is aborted and RTP from the new IP address is ignored as the session is not moved to the STRICT_RTP_LEARN state.
> This behavior was introduced in commit https://github.com/asterisk/asterisk/commit/40de3a12e0caddec0be31aa4ad996c22fc716be5. Reverting that code will however cause Asterisk to segfault on the scenario as described.
> It's not clear to me yet why *session->pending_media_state->topology* is null in this scenario, as opposed to all other scenarios where SDP is updated for an existing session.
> *1
> {code}
> SIP/2.0 183 Session Progress
> v=0
> o=root 1910284739 1910284739 IN IP4 1.2.3.4
> c=IN IP4 1.2.3.4
> {code}
> *2
> {code}
> SIP/2.0 200 OK
> v=0
> o=root 4067 4067 IN IP4 5.6.7.8
> c=IN IP4 5.6.7.8
> {code}



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