[Spce-user] differences between standar calls and VSC call?

[Sergio Serrano]

Hi all,

	I continue review proxy.cfg file to investigate that item, but
I don't understand how kamailio proxy could put advertised IP of
SIPWise inside of SDP(c and o  parameters) only for sw_vsc app and not
for the standar calls. 

Anyone more with something simliar?

Regards,

SergioEl mar, 25-10-2016 a las 11:05 +0200, Sergio Serrano escribió:
> Hi again,
> 
> 	here I am with other extrange case(probably for my limited
> knowledge).
> 
> I have SIPWISE CE confgiured and all standar calls works. But now we
> want to setup VSC and We haven't audio when activate or deactivate a
> service. I launch sngrep and I see next:
> 
> INVITE sip:*72*666666666 at A.A.A.A:59060 SIP/2.0
> .........
> v=0
> o=ATCOM 3686380481 3686380481 IN IP4 87.221.105.155
> s=ATCOM Audio Call
> c=IN IP4 87.221.105.155
> t=0 0
> m=audio 57464 RTP/AVP 8 0 101
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20This invite is passed from LB to Proxy as
> 
> INVITE sip:*72*666666666 at A.A.A.A:59060 SIP/2.0
> .........
> v=0
> o=ATCOM 3686380481 3686380481 IN IP4 87.221.105.155
> s=ATCOM Audio Call
> c=IN IP4 87.221.105.155
> t=0 0
> m=audio 57464 RTP/AVP 8 0 101
> a=sendrecv
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> 
> And this invite is passed from Proxy to sems as
> 
> INVITE sip:*72*666666666 at vsc.local:59060 SIP/2.0
> v=0
> o=ATCOM 3686380481 3686380481 IN IP4 A.A.A.A
> s=ATCOM Audio Call
> c=IN IP4 A.A.A.A
> t=0 0
> m=audio 31756 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> a=rtcp:31757
> 
> This SDP make to rtpengine send audio to A.A.A.A which is the Public
> IP
> of SIPWise.
> 
> Where Can I see to find error?
> 
> Best Regards and thanks for your help,
> 
> Sergio
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