I'm using Asterisk-13.21. I'll check out the code in ast_rtp_interpret but the problem is that I do not have the access to the production environment to recreate this issue.
can anybody suggest any tool to create dummy RTP payloads or some SIP client that can generate real-time text over RTP? On Fri, 31 Jan 2020 at 16:51, Joshua C. Colp <jc...@sangoma.com> wrote: > On Fri, Jan 31, 2020 at 3:06 AM Mohit Dhiman <mohitdhiman...@gmail.com> > wrote: > >> Hi, >> I'm trying to debug a segfault in ast_frdup which happened because of >> the negative datalen of the ast_frame for frame type AST_FRAME_TEXT. >> >> My question is that how an RTP frame in categorized as of type TEXT >> because I can only see two types of RTP payload in network capture (not of >> the time of segfault) >> one is G-711 ulaw and the other is Payload Type 106 (not defined in SDP). >> This Payload 106 is received at the start of the RTP stream and is >> received only once in an RTP stream. >> >> My other question is how the datalen gets calculated for an RTP frame >> and what could be the possible reason for this to come out negative for it >> should never be negative as confirmed by Joshua on Asterisk Community Forum. >> >> It would be a great help if anybody could help me figure this out. >> > > What version of Asterisk is in use? > > Otherwise the code itself that interprets RTP packets is > ast_rtp_interpret[1] in res_rtp_asterisk.c. Adding log messages or reading > that would probably yield information. > > [1] > https://github.com/asterisk/asterisk/blob/master/res/res_rtp_asterisk.c#L6932 > > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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