[Bug 32808] New: tp-sofiasip vs. INVITE without SDP-body

bugzilla-daemon at freedesktop.org bugzilla-daemon at freedesktop.org
Tue Jan 4 03:45:08 CET 2011


           Summary: tp-sofiasip vs. INVITE without SDP-body
           Product: Telepathy
           Version: unspecified
          Platform: Other
        OS/Version: All
            Status: NEW
          Severity: normal
          Priority: medium
         Component: sofiasip
        AssignedTo: mikhail.zabaluev at nokia.com
        ReportedBy: stesie at brokenpipe.de
         QAContact: telepathy-bugs at lists.freedesktop.org


I'm trying to connect my Nokia N900 to an Auerswald COMpact 5020 VoIP PBX,
registration & outgoing calls work like a charm, but I've got a problem with
incoming calls.

The PBX provides INVITEs without an SDP-body to the phone, which in turn
answers 100 & 180 (ringing).  If I click the accept-button the phone sends a

--- LOG excerpt ---
(telepathy-sofiasip:32602): tpsip-DEBUG: _tpsip_media_channel_add_member:
accepting an incoming invite
(telepathy-sofiasip:32602): tpsip-DEBUG: tpsip_media_session_debug: SIP media
session [invite-received  ]: accepting the session
(telepathy-sofiasip:32602): tpsip-WARNING **: priv_session_respond: cannot
respond with a valid SDP answer, were all streams closed?
(telepathy-sofiasip:32602): tpsip-DEBUG: priv_session_rollback: enter
(telepathy-sofiasip:32602): tpsip-DEBUG: tpsip_media_session_terminate: enter
(telepathy-sofiasip:32602): tpsip-DEBUG: tpsip_media_session_terminate: sending
the 480 response to an incoming INVITE
--- full log at: http://paste.q0a.de/m3562

... instead of that I'd expect it to reply a 200 with SDP-answer, but I can't
figure out how to get tp-sofiasip to do that for me :-)

I've tried several variations of ...

      user_sdp = g_string_append (user_sdp, "m=audio 0 RTP/AVP 0\r\n");
      nua_respond (priv->nua_op, SIP_200_OK,
                   SOATAG_USER_SDP_STR (user_sdp->str),

... and the like in the else-part of priv_session_respond, but that doesn't
help much.  Session negotiation doesn't work that way either, probably since a
real port no. must be provided, but how to set these up!?

I have not much of a clue of all the SIP-stuff and really got stuck here :-/

any help appreciated, cheers,


PS: I've tried version 0.6.3 and Git/master

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