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
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