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

[lvl (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27441?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=240254#comment-240254 ] 

lvl edited comment on ASTERISK-27441 at 11/23/17 8:23 AM:
----------------------------------------------------------

Right, I thought you only wanted the pjsip logger part. I'll attach the full debug log, although I'm pretty sure the "Pending topology was NULL for channel" message is the only relevant part. Asterisk is learning RTP when the 183 comes in but won't restart learning after the 200. That's in line with what I can see in the code, because "Pending topology was NULL for channel" indicates that the 200's SDP won't be applied.

A few questions at this stage:

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)
2. Wouldn't it at the least warrant a debug message when this occurs?
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?
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?

I indeed use the default of with-pjproject-bundled.


was (Author: lvl):
Right, I thought you only wanted the pjsip logger part. I'll attached the full debug log, although I'm pretty sure the "Pending topology was NULL for channel" message is the only relevant part. Asterisk is learning RTP when the 183 comes in but won't restart learning after the 200. That's in line with what I can see in the code, because "Pending topology was NULL for channel" indicates that the 200's SDP won't be applied.

A few questions at this stage:

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)
2. Wouldn't it at the least warrant a debug message when this occurs?
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 these?
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?

I indeed use the default of with-pjproject-bundled.

> 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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