CPE SIP Products

 

 

 

 

 

 

 

 

 

Parameter

 

Description

 

 

 

 

 

 

 

 

 

 

 

QoS statistics in SIP

 

Determines whether the device includes call quality of service (QoS)

 

 

Release Call

 

statistics in SIP BYE and SIP 200 OK response to BYE, using the

 

 

[QoSStatistics]

 

proprietary SIP header, X-RTP-Stat.

 

 

 

 

 

 

[0] = Disable (default)

 

 

 

 

 

 

[1] = Enable

 

 

 

 

 

 

The X-RTP-Stat header provides the following statistics:

 

 

 

 

Number of received and sent voice packets

 

 

 

 

Number of received and sent voice octets

 

 

 

 

Received packet loss, jitter (in ms), and latency (in ms)

 

 

 

 

The X-RTP-Stat header contains the following fields:

 

 

 

 

PS=<voice packets sent>

 

 

 

 

 

 

OS=<voice octets sent>

 

 

 

 

 

 

PR=<voice packets received>

 

 

 

 

 

 

OR=<voice octets received>

 

 

 

 

 

 

PL=<receive packet loss>

 

 

 

 

 

 

JI=<jitter in ms>

 

 

 

 

 

 

LA=<latency in ms>

 

 

 

 

 

 

Below is an example of the X-RTP-Stat header in a SIP BYE

 

 

 

 

message:

 

 

 

 

 

 

 

 

 

 

 

 

 

BYE sip:302@10.33.4.125 SIP/2.0

 

 

 

 

 

 

Via: SIP/2.0/UDP 10.33.4.126;branch=z9hG4bKac2127550866

 

 

 

 

 

Max-Forwards: 70

 

 

 

 

 

 

From: <sip:401@10.33.4.126;user=phone>;tag=1c2113553324

 

 

 

 

 

To: <sip:302@company.com>;tag=1c991751121

 

 

 

 

 

Call-ID: 991750671245200001912@10.33.4.125

 

 

 

 

 

CSeq: 1 BYE

 

 

 

 

 

 

X-RTP-Stat:

 

 

 

 

 

 

PS=207;OS=49680;;PR=314;OR=50240;PL=0;JI=600;LA=40;

 

 

 

 

 

Supported: em,timer,replaces,path,resource-priority

 

 

 

 

 

Allow:

 

 

 

 

 

 

REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REF

 

 

 

 

 

ER,INFO,SUBSCRIBE,UPDATE

 

 

 

 

 

 

User-Agent: Sip-Gateway-/v.6.2A.008.006

 

 

 

 

 

Reason: Q.850 ;cause=16 ;text="local"

 

 

 

 

 

Content-Length: 0

 

 

 

 

 

 

 

 

 

 

 

RTP-Only Mode

 

 

 

 

 

 

RTP Only Mode

 

Enables the device to start sending and/or receiving RTP packets to

 

 

[RTPOnlyMode]

 

and from remote endpoints without the need to establish a SIP

 

 

 

 

session. The remote IP address is determined according to the

 

 

 

 

Outbound IP Routing table. The port is the same port as the local RTP

 

 

 

 

port (configured by the parameter BaseUDPPort and the channel on

 

 

 

 

which the call is received).

 

 

 

 

 

 

[0] Disable (default)

 

 

 

 

 

 

[1] Transmit & Receive = Send and receive RTP

 

 

 

 

[2] Transmit Only= Send RTP only

 

 

 

 

 

 

[3] Receive Only= Receive RTP only

 

 

 

 

 

 

Notes:

 

 

 

 

 

 

To configure the RTP Only mode per trunk, use the

 

 

 

 

RTPOnlyModeForTrunk_ID.

 

 

 

 

 

 

If per trunk configuration (using RTPOnlyModeForTrunk) is set to a

 

 

 

 

 

 

 

 

 

SIP Release Notes

88

Document #: LTRT-26901

 

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AudioControl VERSION 6.2 manual Parameter Description, QoSStatistics, RTP-Only Mode, RTPOnlyMode