The problem is the title of this thread is “Zulu client does not use end-to-end rtp”. His requirements are specifically Zulu (which uses WebRTC) and RTP. Utilizing Kamailio doesn’t do anything for him, in fact it negates the whole point of this thread which is “zulu”
Then I guess the answer is “yes, that’s right; it does not”
How else can I be of service? 
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Well perhaps true, that was his original requirements, but because such requirements can’t currently be fulfilled by Zulu so that path is self negated, his apparent “real” requirement is just webrtc without a PBX, the thread dynamically changed. . . . No?
No the point is not that I need just WebRTC. And I do use a lot of PBX features offered by Asterisk , but a very limited part that obstruct Direct Media (like recording) I use only in exceptional cases.
As said with SIP and non encrypted RTP I can use Direct Media and still benefit of a lot of Asterisk functionality.
Then comes the point of softclients - also an ongoing trend to replace hardphones - and I’ve been using Counterpath BRIA for a while, but the cooperation with FreePBX has been abandoned, and I wasn’t too happy after all about their multi-platform compatibility. That’s why I tried the Zulu Softclient and my first impressions were promising. It’s just the missing Direct Media that’s holding me off.
So if there is no chance in getting this with Zulu/WebRTC, my only choice is to look for another SIP based softclient which offers multi-platform support and integrates with FreePBX Chat and Presence. Any suggestions would be very welcome !
So we have beat up on why zulu/astersik can’t do that for ever?,It still can’t, no matter how much you want it to, so just put your calls to a proxy that can, why is that a problem? please understand that there will NEVER be a b2bua that can do what you want.
it does it for SIP , so if it can’t for WebRTC, then I need to stick with SIP
Sip is NOT webrtc , no matter how much you want it to be it just won’t be, sorry a SIP proxy can arrange that, a b2bua just can’t
SIP has a REINVITE or UPDATE mechanism for “bypassing” the B2BUA for media. WebRTC does no seem to provide that, but should provide end-to-end RTP per default, but not with Asterisk.
Even when it would, then of course remains the case where a SIP endpoint calls a WebRTC client. But even that should be possible, because in the end the media is for both RTP or SRTP - it’s just a matter of agreeing a common codec…
You are absolutely right, BUT Asterisk just can’t get out of the call path, it is never going to be a Proxy, as everyone has told you (repeatedly) , it is B2BUA, I suggest you just stop your dog chewing that bone.
Again this is easy when its RTP. Once your SRTP this is not so easy and not sure why you want to fight with us on this. Take it up with Digium please we have no say in how Asterisk works. We just work with the tools we have. Take it to Digium and fight your battle with them or hire someone to add the feature and submit it to Digium.
Almost every hosted provider I know of out their proxies media and yet you seem to be the only one who doesn’t like this yes companies doing 100M plus in revenue have no issues selling such services.
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