Hi Ryan,
yeah, this happens because OpenSIPS applies all the changes at the end,
when the message is about to be sent out. As a side effect, when sending
the SDP to rtpengine, opensips does not see its own previous changes
over the body (changes are still pending).
Usually there are easy workarounds for this, but in this case it looks
like bug to me. Could you please open a bug report the the github tracker.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 04/09/2018 05:22 PM, Esty, Ryan wrote:
Hi opensips list,
First some background I’m trying to use opensips as a webrtc proxy. I
found out that the packets for the invite going to my sip server are
too big for my sip server. It doesn’t like packets to be over 4000
bytes. I’m trying to take what I can out of the sip packets like codes
I know the other side can’t do. First codec stripping works but only
with the audio codecs. If I try to strip a video codec the packet gets
mangled. This is probably a bug in rtpengine and not opensips. I was
hoping if anyone has any idea how I might get my invite packets
smaller? The webrtc side is generating ssrc lines in my sdp. I’m
trying to strip them out but I’m not sure if rtpengine is putting them
back in or not. Before my rtpengine_offer I do a
replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has all the
ssrc lines in it.
Ryan
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users