[asterisk-bugs] [JIRA] (ASTERISK-29929) res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions

[Boris P. Korzun (JIRA)]

Boris P. Korzun created ASTERISK-29929:
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             Summary: res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions
                 Key: ASTERISK-29929
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29929
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_pjsip_sdp_rtp
    Affects Versions: 18.9.0
            Reporter: Boris P. Korzun


There's an unresolved bug ASTERISK-26689.
It's found out that RTC timer has been starting on first media negotiation in SIP Dialog (ex, after receiving 183 with SDP). But it isn't correct to compute RTP timeout from first media negotiation. It should be do only after channel establishment.
Also it's not needed to check RTP while direct media is used (Asterisk doesn't took part in the media session).

According to [SIP 183 Session Progress Message|https://www.ietf.org/proceedings/46/I-D/draft-ietf-sip-183-00.txt] before called user agent send OK or ACK there is one way SDP.

I think to make rtp_timeout option in chan_pjsip consistent to rtptimeout option in chan_sip.



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