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Yes, this is possible using RTCP.

If you turn on the setting RTCP support in the "advanced" webpage from the Web_UI of the phone, the phone will send standard RTCP Sender Reports to the other phone periodically. They contain the relevant call information including packet loss, fraction lost, jitter, timestamps info. and other information.

Example

PUBLISH SIP/2.0 

From: <sip:446@intern.snom.de>;tag=45hkui59ns
To: <intern.snom.de>;tag=nohhk4xu21
Call-ID: 3c26a8de500f-12ct7zov3kjs
CSeq: 3 PUBLISH
Max-Forwards: 70
Contact: <sip:446@192.168.5.251:2060;transport=tls;line=w2wuvhk9>;reg-id=1
Event: vq-rtcpxr
Accept: application/sdp, message/sipfrag
Content-Type: application/vq-rtcpxr
Content-Length: 832

VQSessionReport
LocalMetrics:
Timestamps:START=2010-02-17T13:59:42Z STOP=2010-02-17T13:59:46Z
SessionDesc:PT=0 PD=G.711U PPS=50 SSUP=off
CallID:3c26a8de500f-12ct7zov3kjs
x-UserAgent:snom360/8.2.sf
FromID:<sip:446@intern.snom.de>
ToID:<sip:447@intern.snom.de;user=phone>
LocalAddr:IP=192.168.5.251 PORT=62754 SSRC=0xCBE3450E
RemoteAddr:IP=192.168.0.233 PORT=54018 SSRC=0xB80B52F3
DialogID:3c26a8de500f-12ct7zov3kjs;to-tag=866ed0cf03;from-tag=45hkui59ns
x-SIPmetrics:SVA=RG SRD=310 SFC=0
x-SIPterm:SDC=OK
JitterBuffer:JBA=0 JBR=0 JBN=0 JBM=0 JBX=65535
PacketLoss:NLR=0.0 JDR=0.0
BurstGapLoss:BLD=0.0 BD=0 GLD=0.0 GD=6569 GMIN=16
Delay:RTD=0 ESD=0 IAJ=4
RemoteMetrics:
JitterBuffer:JBA=0 JBR=0 JBN=0 JBM=0 JBX=0
PacketLoss:NLR=0.0 JDR=0.0
BurstGapLoss:BLD=0.0 BD=0 GLD=0.0 GD=4677 GMIN=16
Delay:RTD=0 ESD=0 IAJ=2


For a complete setup guide, please have a look at our feature. support_rtcp


Further Information


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