On Mon, 21. Dec 16:35, Aman Karmani wrote: > From: Aman Karmani <a...@tmm1.net> > > The SAT>IP protocol[1] is similar to RTSP. However SAT>IP servers > are assumed to speak only MP2T, so DESCRIBE is not used in the same > way. When no streams are active, DESCRIBE will return 404 according > to the spec (see section 3.5.7). When streams are active, DESCRIBE > will return a list of all current streams along with information > about their signal strengths. > > Previously, attemping to use ffmpeg with a rtsp:// url that points > to a SAT>IP server would work with some devices and fail due to 404 > response on others. Further, if the SAT>IP server was already > streaming, ffmpeg would incorrectly consume the DESCRIBE SDP response > and join an existing tuner instead of requesting a new session with > the URL provided by the user. These issues have been noted by many > users across the internet[2][3]. > > This commit adds proper spec-compliant support for SAT>IP, including: > > - support for the satip:// psuedo-protocol > - avoiding the use of DESCRIBE > - parsing and consuming the com.ses.streamID response header > - using "Transport: RTP/AVP;unicast" because the optional "/UDP" > suffix confuses some servers > > [1] > https://www.satip.info/sites/satip/files/resource/satip_specification_version_1_2_2.pdf > [2] > https://stackoverflow.com/questions/61194344/does-ffmpeg-violate-the-satip-specification-describe-syntax > [3] https://github.com/kodi-pvr/pvr.iptvsimple/issues/196 > --- > libavformat/rtsp.c | 52 ++++++++++++++++++++++++++++++++++++++----- > libavformat/rtsp.h | 6 +++++ > libavformat/rtspdec.c | 1 + > 3 files changed, 53 insertions(+), 6 deletions(-) > > diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c > index c7ffa07d9e..4a863dbac3 100644 > --- a/libavformat/rtsp.c > +++ b/libavformat/rtsp.c > @@ -252,6 +252,24 @@ static void finalize_rtp_handler_init(AVFormatContext > *s, RTSPStream *rtsp_st, > } > } > > +static int init_satip_stream(AVFormatContext *s) > +{ > + RTSPState *rt = s->priv_data; > + RTSPStream *rtsp_st; > + rtsp_st = av_mallocz(sizeof(RTSPStream)); > + if (!rtsp_st) > + return AVERROR(ENOMEM); > + rtsp_st->stream_index = -1; > + dynarray_add(&rt->rtsp_streams, > + &rt->nb_rtsp_streams, rtsp_st); > + av_strlcpy(rtsp_st->control_url, > + rt->control_uri, sizeof(rtsp_st->control_url)); > + rtsp_st->sdp_payload_type = 33; // MP2T > + init_rtp_handler(&ff_mpegts_dynamic_handler, rtsp_st, NULL); > + finalize_rtp_handler_init(s, rtsp_st, NULL); > + return 0; > +} > + > /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] > */ > static int sdp_parse_rtpmap(AVFormatContext *s, > AVStream *st, RTSPStream *rtsp_st, > @@ -1116,6 +1134,9 @@ void ff_rtsp_parse_line(AVFormatContext *s, > } else if (av_stristart(p, "Content-Type:", &p)) { > p += strspn(p, SPACE_CHARS); > av_strlcpy(reply->content_type, p, sizeof(reply->content_type)); > + } else if (av_stristart(p, "com.ses.streamID:", &p)) { > + p += strspn(p, SPACE_CHARS); > + av_strlcpy(reply->stream_id, p, sizeof(reply->stream_id)); > } > } > > @@ -1495,8 +1516,10 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, > const char *host, int port, > rtp_opened: > port = ff_rtp_get_local_rtp_port(rtsp_st->rtp_handle); > have_port:
> - snprintf(transport, sizeof(transport) - 1, > - "%s/UDP;", trans_pref); > + av_strlcat(transport, trans_pref, sizeof(transport)); transport needs to be initialized to an empty string. Otherwise av_strlcat will keep all the random bytes before the first NULL. -- Andriy _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
