Trableshooting J100

[Shooroop]

Good day, there is such a problem.
From time to time, about every 15 calls are made with a troubleshooting. The called or caller does not hear the incoming RTP stream. This happens in any combination (with an outgoing or incoming call, inaudibly or dialed or dialed). By the method of scientific poke, we found out that there is a situation with phones using the SIP protocol, i.e. between sip-sip, sip-2420, sip-analog. By switching the phones to tcp, it was possible to remove the dump of calls. It shows two RTP streams, one for example is correct when a person speaks and packets are sent and a problematic one that does not go, packets are not sent, with the exception of one. In all settings there are Direkt ip to ip. From the general dump, it is immediately clear that the phone, as if by itself, stops sending packets. Aura firmware is the latest, 8.1.3.3, phones and gateways too. SLA Mon does not give any sense either, it is a program for tracking the lower levels of the OSI model. There is an assumption that this is an application-level error, software or some kind of setup that no one knows about.

 

[kyle555]

What language did you use for Google translate?

Good job troubleshooting.

Are the J100 phones connected directly to SM or through an SBC?

Look at the good calls and bad calls. In the SIP messages - usually the INVITE and 200OK you'll see a body of SDP at the end. It defines how media is to be setup.

The c=10.112.23.44 and the m=audio 6064 in the example below means "send me audio on 10.112.23.44:6064". Check in ACLs/firewalls that the ports negotiated for bad calls aren't blocked.

In the J100 phone's 46xxsettings.txt file there are 2 parameters to define what RTP ports to use. The examplebelow would force the phone to use ports 40001-44096. In CM, it's in the ip-network-region configuration for gateways and H323 phones where you define the port ranges for those devices. If you use Media Server, it's in the web gui for that specific type of device.

SET RTP_PORT_LOW 40001
SET RTP_PORT_RANGE 4096

v=0
o=- 1543832047 2 IN IP4 10.112.23.41
s=-
c=IN IP4 10.112.23.44
b=AS:64
t=0 0
m=audio 6064 RTP/AVP 8 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=ptime:20