[Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

[H Yavari]

Hi,
I can configure RTP setting for peers but this is not work for me. For example I have an asterisk as a peer and I selected RTP/AVP. but SDP in calls from webrtc clients to peer is : UDP/TLS/RTP/SAVP. So asterisk reject the calls.

But about dynamically: 
why I can't make a call between sip extensions and webrtc clients that are registered on ngcp? I found in log that SDP for callee doesn't change automatically.

Regards,H.Yavari
      From: Andrew Pogrebennyk <apogrebennyk at sipwise.com>
 
   
On 04/14/2015 12:55 PM, H Yavari wrote:
> I have some problem with new version (3.8,1).
> webrtc clients to webrtc clients calls are ok when clients are same. I
> mean when caller and callee are on SIPmpl5 or Sip.js calls are ok.
> I think the reason is: in this version, we can't select more than one
> policy for RTP for a domain. So webrtc client <---> sip client calls are
> not possible too.
> 
> In 3.7 version, we could force RTP/AVP for peer servers. but in 3.8.1
> this part is changed and we can't do this. So WebRTC calls to SIP legacy
> peers are not possible (SDP incompatible) .

I'm not sure if I understand you, can't you set the transport protocol
on the peer in the mr3.8.1? The only difference between two releases,
from my point of view is that in mr3.8.1 you have one preference instead
of two (rtcp_feedback has been combined with rtp transcoding).

> It is possible that NGCP make decision about SDP dynamically ?

We kind of do this already, because if callee is on the websocket we
default to UDP/TLS/RTP/SAVPF.



Andrew


   
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