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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