Sip trace:

For your understanding - im using kamailio as registrar there. FS as media and Services server
so call goes:

WebRTC client A -> (websockets) -> Kamailio -> (udp) -> FreeSwitch
|__________________Leg A__________________________________|


FreeSwitch -> (udp) -> Kamailio -> (websockets) -> WebRTC client B
|__________________Leg B__________________________________|


http://pastebin.com/swBDJnYG


2016-09-13 9:13 GMT+03:00 Yuriy Gorlichenko <ovoshlook@gmail.com>:
I will make sip trace little bit later (in 2-3 hours from now).
And yes.
For now it is a simple proxy call from one A to B without any changes. 
And the SDP all ok. B leg has 
a=video
and its staff

At the next step i need to generate B leg by myself from freeswitch after some handling leg A. So i suppose that settings for now not enought for this scenario. am i right?


2016-09-13 8:42 GMT+03:00 Mirko Brankovic <mirkobrankovic@gmail.com>:

I'm expecting to see b side without video capabilities.
So do the sofia global sip trace on and see the invite leaving fs box. Check if sdp has both audio and video m flags, and what was in 200ok sdp back from b side.


On Sep 12, 2016 20:28, "Dave Horton" <daveh@beachdognet.com> wrote:
Thanks, and a SIP trace showing the call, and the failed negotiation for video?

To be clear, is this a simple bridging scenario (you have an inbound call offering video and you want to bridge it to an outbound B leg which also offers video)?


On Sep 12, 2016, at 10:42 AM, Yuriy Gorlichenko <ovoshlook@gmail.com> wrote:

http://pastebin.com/e4AXe3xC

But i suppose this helped.
Also codecs mached

Am i will be right if i say that 1.6 supports VP8?
I just asking because not found any mod_vpx or something like thais at the mod dir.

2016-09-12 17:35 GMT+03:00 Dave Horton <daveh@beachdognet.com>:
There are a bunch of params that can impact SDP negotiation.  I think it would be most helpful if you could pastebin the sip profile settings and then an actual trace of a call

On Sep 12, 2016, at 9:37 AM, Yuriy Gorlichenko <ovoshlook@gmail.com> wrote:

Yep. that in it. 

freeswitch/vars.xml
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/>

freeswitch/sip_profiles/external.xml

<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>


I making call as

session:execute("bridge","sofia/external/"..session:getVariable("destination_number").."@my.sip.pro.xy")

2016-09-12 16:24 GMT+03:00 Mirko Brankovic <mirkobrankovic@gmail.com>:
make sure you have video codecs enabled in vars.conf for outbound codec list

On Mon, Sep 12, 2016 at 2:57 PM, Yuriy Gorlichenko <ovoshlook@gmail.com> wrote:
Hi. Googled documenttion about it
where i can turn it on?
For now icoming INVITE goes with Video but then FS omakes outgoing call without Video Paert at the SDP.

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--
Regards,
Mirko

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