[FFmpeg-user] rtp source stopped working upgrading ffmpeg from3.0.0 to 3.1.1

[Mark Hassman]

> -----Original Message-----
> From: ffmpeg-user [mailto:ffmpeg-user-bounces at ffmpeg.org] On 
> Behalf Of Carl Eugen Hoyos
> Sent: Thursday, July 07, 2016 2:44 AM
> To: ffmpeg-user at ffmpeg.org
> Subject: Re: [FFmpeg-user] rtp source stopped working 
> upgrading ffmpeg from3.0.0 to 3.1.1
> 
> Mark Hassman <mark <at> hassman.org> writes:
> 
> > Reading source input from rtp no longer works.
> 
> Please either explain how I can reproduce (explain it as if I 
> had never heard of rtp) or tell us which change introduced the issue.
> 
> Thank you, Carl Eugen


Hi,

Apologies on delayed response..

The rtp stream is generated through an automated workflow.
I've created a custom test page capable of generating it:
https://dev01.privatecircle.com:8091/ffmpeg/.. add your local ip/port and
click start, rtp will be streamed to you. The test page leverages webrtc and
works in chrome/firefox.

fyi.. i increased my debugging level, issue is locating i-frame in h.264
input.. but, this same rtp stream works fine with ffmpeg v3.0.

[h264 @ 0x22c18c0] non-existing PPS 2 referenced
[h264 @ 0x22c18c0] nal_unit_type: 1, nal_ref_idc: 3
[h264 @ 0x22c18c0] non-existing PPS 2 referenced
[h264 @ 0x22c18c0] decode_slice_header error
[h264 @ 0x22c18c0] no frame!

Thoughts?
Thnx!


-Mark