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



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