#1457 closed defect (fixed)
Assertion when incoming reoffer contains no media (thanks Ashraf Jaddo for the report) — at Version 2
Reported by: | nanang | Owned by: | nanang |
---|---|---|---|
Priority: | normal | Milestone: | release-1.14 |
Component: | pjmedia | Version: | 1.x-branch |
Keywords: | Cc: | ||
Backport to 1.x milestone: | Backported: |
Description (last modified by nanang)
Original report can be found here.
I am using PJSIP 1.12 with iPhone 5.0.. I am getting “Assertion failed: (sdp_remote && m_rem), function transport_encode_sdp, file ../src/pjmedia/transport_srtp.c, line 1299.” every time I try to call certain numbers.. ... the problem is when I call from iPhone to a PBX extension that is connected to the service.. and I I get the above error, to be specific the code crash only when I pick up the call from that extension.. Snippet log: ... (normal outgoing initial INVITE until ACK ..) ... 15:31:31.393 pjsua_core.c RX 703 bytes Request msg UPDATE/cseq=102 (rdata0x9061bc) from tcp 10.50.0.4:5060: UPDATE sip:77999991@10.50.1.10:5060;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TCP 10.50.0.4:5060;branch=z9hG4bK4b370dd7;rport Max-Forwards: 70 From: sip:9874@10.50.0.4;tag=as5a9c25b2 To: sip:77999991@10.50.0.4;tag=PxbkZXr7smjwglhT87678rIypjrr0tln Contact: <sip:9874@10.50.0.4:5060;transport=TCP> Call-ID: F3a8f5vZu0f6JEJFmlN5vIagy-P1De7h CSeq: 102 UPDATE User-Agent: TEST Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 102 v=0 o=root 2007926168 2007926170 IN IP4 172.16.201.225 s=TEST c=IN IP4 172.16.201.225 t=0 0
Such reoffer should be just simply rejected and any existing session won't get updated.
Change History (2)
comment:1 Changed 13 years ago by nanang
- Resolution set to fixed
- Status changed from new to closed
comment:2 Changed 13 years ago by nanang
- Description modified (diff)
Note: See
TracTickets for help on using
tickets.
(In [3962]) Fix #1457: removed check for remote SDP media count before calling find_audio_index() in pjsua_media_channel_create_sdp(), so find_audio_index() will also verify the media count in the remote SDP.