[asterisk-bugs] [JIRA] (ASTERISK-27441) No audio after early media from external endpoint

[Richard Mudgett (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27441?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=240264#comment-240264 ] 

Richard Mudgett commented on ASTERISK-27441:
--------------------------------------------

{quote}
1. Where is the code that ignores an SDP when the version number is not higher than the previous version number? (Actually, it's not completely ignored because at least _handle_negotiated_sdp_ is called)
{quote}
I think that is checked in pjproject.  Another place it could be checked is in res_pjsip_session.c.
{quote}
2. Wouldn't it at the least warrant a debug message when this occurs?
{quote}
Yes.  A debug message would be a good thing.
{quote}
3. I see some relevant PJ_LOG statements in pjsip's code but these messages don't appear in asterisk's debug log. Is it possible to see them?
{quote}
See CLI "pjproject set log level" and {{pjproject.conf.sample}}.  Beware that those are pjproject's log levels that are routed through Asterisk's logging system.  Also level 5 and 6 output a rediculous amount and is rarely useful.
{quote}
4. Chan_sip has an option to gracefully deal with broken clients in this exact scenario: ignoresdpversion. Would it be possible to add this in Chan_Pjsip as well?
{quote}
See issue ASTERISK-27309 about about getting this option incorporated into chan_pjsip.

> No audio after early media from external endpoint
> -------------------------------------------------
>
>                 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
>
>
> 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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