VSOL Certified ICT Professional-Access

V2.0

VoIP(SIP) Services
and Troubleshooting

VSOL | PON/CPE Manufacturer &

FTTX/MBB Solution Provider
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 01 SIP Protocol
CONTENTS

02 SIP Application

03 VoIP Troubleshooting

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 01 Sip protocol
Welcome to VSOL VCIP Training

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 Objectives

Upon completion of this course, you will be able to:

Understand the functions and principles of the SIP protocol.
Understand the interaction process of SIP messages in registration.
Understand the interaction process of SIP messages during a call.

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 Session Initiation
Protocol
n The Session Initiation Protocol(SIP)is an application layer control
protocol for multimedia communication on the IP network.It is used to
create,modify,and terminate session processes of one or more participants.

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 SIP Functions

The functions of the protocol are as follows:

∎ Subscriber positioning: Determine the location of a terminal subscriber participating in

communication.
∎ Communication capability negotiation: Determine the media type and parameters of communication.
∎ Obtaining subscriber intention: Determine whether the callee is willing to participate in a
communication
∎ Settingup a calt Ring to the called part, and determine the call parameters of the caller and
callee.
∎ Call processing and controt Include call redirection, transfer, and termination.

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 SIP Principles-Basic Network Model

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 SIP registration
Each time when the SIP client is powered on, the SIP client needs to register with the SIP server.
When the IP address of the SIP client changes, the SIP client also needs to register with the SIP
server again. The registration information must be updated periodically.
The following takes the registration of a SIP phone with the SIP server as an example to describe
the registration of a SIP subscriber.

SIP Phone SIP Server

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 SIP registration

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 SIP Call Flow between SIP Entities(1)
n Create a SIP call

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 SIP Call Flow between SIP Entities(2)
n conversation and Release call

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 SIP Call Flow between SIP Entities(2)

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 02 SIP Application

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 Objectives

n Understand the SIP configuration of ONU web.

.

Training
Objectives
n Understand the SIP configuration of GPON/EPON OLT.

n Understand the VoIP service.

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 VoIP Service Networking
ONT voice service:
The ONU connects to a common phone through its
POTS port, converts analog signals into
digital signals through internal digital-to-
analog conversion, and packs protocols. The
ONU supports SIP protocol. The ONU can also
be connected to the SIP phone of the user
through the Ethernet port. In this case, the
SIP phone uses the SIP protocol to complete
the packaging of user data, and transparently
transmits the data to the upper-layer
softswitch for processing through the ONT and
OLT.

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 VoIP Service Configuration of ONU

Step 1: create a VoIP WAN connection

1. Select network mode

2. Fill in the VLAN ID and priority

corresponding to VoIP Wan

3. Select VoIP service mode

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 VoIP Service Configuration of ONU

Step 2: Fill in the SIP server info

1. Fill in register server address

2. Fill in proxy server address

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 VoIP Service Configuration of ONU

Step 3: Fill in the SIP account info

1. Fill in user phone number
2. Fill in SIP account password

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 VoIP Service Configuration of ONU

If SIP registration is successful, "Registerd" will be displayed here

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 VoIP Service Configuration of GPON OLT

Step 1: create a VoIP WAN connection

1. Select network mode and fill in network

setting

2. Fill in the VLAN ID and priority

corresponding to VoIP Wan

3. Select VoIP service mode

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 VoIP Service Configuration of GPON OLT

Step 2: Fill in the SIP server info

1. Fill in register server address

2. Fill in proxy server address

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 VoIP Service Configuration of GPON OLT

Step 3: Fill in the SIP account info

1. Fill in user phone number
2. Fill in SIP account and password

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 VoIP Service Configuration of GPON OLT

If SIP registration is successful, "Idle"

will be displayed here

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 VoIP Service Configuration of EPON OLT

Step 1: create a VoIP WAN connection

1. Select network mode and fill in network

setting

2. Fill in the VLAN ID and priority

corresponding to VoIP Wan

3. Select VoIP service mode

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 VoIP Service Configuration of EPON OLT

Step 2: Fill in the SIP server info

1. Fill in register server address

2. Fill in proxy server address

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 VoIP Service Configuration of EPON OLT

Step 3: Fill in the SIP account info

1. Fill in user phone number
2. Fill in SIP account and password

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 VoIP Service Configuration of EPON OLT

If SIP registration is successful, "Idle"

will be displayed here

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 Introduction of VoIP Server
After knowing how to configure VoIP Services,let’s learn about the functions of VoIP

T38 fax mode is a way of fax, If you have no

choice, the default is T30 fax

The way of sending number for caller ID, You

can choose what you need according to local
standards

Ringtone mode, you can choose the ringtone you want

according to your region, including dial tone, busy
tone, ring tone and so on

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 Introduction of VoIP Server

Time of flash detection,If it

is not in this range, it will
not be recognized as flashover

Time of tone duration,

You can customize the
playing time of the tone

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 Introduction of VoIP Server

Heartbeat mode can be enabled on here, including active

mode and passive mode. When SIP server requests to keep
heartbeat, this switch needs to be enable.

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 Introduction of VoIP Server

You can configure the digimap here.

The letter "x" is used as a wildcard, designating any event corresponding to symbols in the
range "0"-"9".
The dot symbol "." stands for zero or more repetitions of the event selector (such wildcard
and digits "0" through "9" and letters "E"“F”) that precedes it.
The letter “E” is button “✱” , The letter “E” is button “#”

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 Introduction of VoIP Server
Call waiting is a telephone
service that sends you a signal
if another call arrives while
youWhen
are you
already
pick on
up the
the phone.
phone,
will dial the designated number
automatically

Call forwarding includes unconditional

forwarding, busy forwarding and no answer
forwarding

When you want to use call forwarding, you

have to enable it,

Attend call transfer includes

consultation transfer and blind transfer
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 03 VoIP Trouble Shooting

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 Objectives

Upon completion of this course, you will be able to:

Understand the VoIP service trouble
Understand troubleshooting methods in different FTTx scenarios.

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 3.1 Data traffic Mirror and Capture
First of all, we will introduce a common debugging method, ONU data mirroring and packet capture.
Realtek series ONU's data image and Wireshark tool capture can record the communication data in the engineering
problem environment, and provide it to senior technical support personnel or R&D personnel to analyze problems.
Please understand this method. We will often use this method later。

The following describes the steps of data mirroring and packet capturing (taking HG323DAC as an example)

Step1:Open Lan side Telent option

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 3.1 Data traffic Mirror and Capture
Step2:Telnet to ONU , and type these commands in the following:

After input the username/password, then type command: enterlinuxshell enter shell mode, type command: diag enter
debug mode of onu.
This command: port get status port all will list the onu current port list, then easy to know which port id to make m
different ONU model , there is different port id list.
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Like this example: make mirror from PON to Lan1 both direction data traffic.
 3.1 Data traffic Mirror and Capture
Step3:Start/Stop capture and save:

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 3.2 Registeration failure
n phenomenon：When you fill in the SIP registration parameters ，the VoIP
registration indicator does not work

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 3.2 Registration failure
Analysis:

Case 1: Check the registration status of ONU, If the registration status is “Registered”, we
think the phone indicator was broken.

Solution:

It's just light problem, it doesn't affect the ONU using，or you can replace the phone indicator light.

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 3.2 Registration failure

Case 2: Check the registration status of ONU, If the registration status is “Registering”,
need to continue to check.

Solution:

step 1 : Check whether the registration information is correct

step 2 : Check whether the VoIP wan connection is normal
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 3.2 Registration failure

If the WAN link is normal, as shown in the

left figure, there can be a normal Ping value.

If the WAN link is abnormal, we need to check

whether the WAN connection configuration
information is correct

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 3.2 Registration failure
Case 3: Check the registration status of ONU, If the registration status is “Registering”
and the number has a suffix,the suffix has a special meaning and cannot be placed inside the
number

Solution:

Remove the suffix from the number

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 3.2 Registration failure
If the ONU sip number does not have a configured suffix, the ONU will send a SIP
registration message, in which you can view the SIP number information,If the ONU
SIP number is configured with a suffix, it will not send SIP registration messages.
So,the solution is removing the suffix from the number.

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 3.2 Registration failure
Failed Case 1 : VoIP VLAN is incorrect

If the VoIP Wan uses an incorrect VLAN, the VoIP Wan cannot transmit data normally, resulting in
registration failure. At this time, you need to confirm whether the VLAN information filled in is
correct.
You can ask the VoIP service provider, or you can check the packet by yourself.

Please refer to 3.1 Mirror to capture the VoIP packets on ONU

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 3.2 Registration failure
This is the correct packet,both sides use the same VLAN for communication

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 3.2 Registration failure
Failed Case 2: There is a problem with the SIP server.

the SIP service of the server is not enabled or server system failure. it can see the
destination port cannot be reached

This case please contact the VoIP service provider

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 3.2 Registration failure
Failed Case 3: The registration port of the server is incorrect.
e.g. the server requires registration port is 5061, but 5060 is used to ONU initiate registration

this case , you need to change

the SIP registration port

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 3.2 Registration failure

The above is a normal SIP registration packet capture, you can compare your packet capture to check which
step is abnormal. Or you can provide it to VSOL technical support team for analysis.

ONU SIP Server

Successfully registered.pcapng

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 3.2 Registration failure
Failed Case 4 :The SIP 403 error,the account is incorrect,it may be account error or
password error

This case, please confirm your registration information again.

If you have been unable to register successfully, please contact your VoIP service provider.

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 3.3 No dial tone when off hook
n phenomenon：When you off hook the telephone, you can’t hear dial tone

n Analysis:
First, check whether the port status of VoIP is registered. If the registration is
“Registering”. please refer to 3.1 for check.If the registration status is “Registered”, but
there is no dial tone, please follow the steps below

Solution:

case 1、Check if you can see the telephone indicator light is on when off hook (As
shown in the picture) . If not, please replace the telephone line or telephone for test.
When you replace the telephone or line, you still can't see the telephone indicator
light on. we judge that the Phone port is faulty.
Please contact the ONU supplier for check.
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 3.3 No dial tone when off hook
Case 2、If the telephone indicator light is on, but you still can't hear the dial tone, please
enable the ONU debugging for check.

Step1:Open Lan side Telent option

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 3.3 No dial tone when off hook
Step2:Telnet to ONU , and type these commands in the following:

After input the username/password, then type command: enterlinuxshell enter shell mode, type command: diag enter
debug mode of onu.
This command: echo t > /var/run/solar_control.fifo
The function of this command is to print the action detection of the telephone
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 3.3 No dial tone when off hook
Step3: Check the difference between your printed information and this picture, Or you
can provide it to VSOL technical support team.

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 3.4 Ring of the phone is not from your area
nphenomenon：When you use the VoIP phone, you find that the dial tone, ring tone and ring
back tone are not consistent with your area.

n Analysis:
You can choose the ring tone standard you need on the ONU web

Solution:

login ONU web , choose the regional ring tone standard you need.

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 3.4 Ring of the phone is not from your area

Choose the regional ring tone

standard you need

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 3.5 ONU can make calls but cannot be called
nphenomenon：When you use the VoIP phone, you find that make calls successfully but can not
be called

n Analysis:
ONU does not send heartbeat packets, MAC address is aging, and the server does not know where
to reply

Solution:

login ONU web , configure heartbeat cycle and maintain heartbeat

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 3.5 ONU can make calls but cannot be called

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 3.6 Can’t hear other side voice in a call
n phenomenon：When you make a call, the called answers, but you can't hear his voices

n Analysis:

Case 1: The two ONUs are under the same OLT PON port

Solution:

Onus under the same PON port are isolated by default, you need to
enable the P2P function or proxy function of PON.

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 3.6 Can’t hear other side voice in a call

EPON OLT P2P function configuration

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GPON OLT P2P function configuration
 3.6 Can’t hear other side voice in a call
Case 2:The two onus use different voice codes

During the use of VoIP phone, We don't know what voice code is used by the other party.
If the voice code of both parties is different, it will cause the problem that both
parties can't hear the voice.

By default, VSOL ONU will automatically negotiate the other party's voice code, but sometimes, The
calling device asks the VSOL ONU to negotiate one code, but it sends another code. Two different
codes in a call cause problems.

Let me give an example to illustrate this problem;

Step 1: Please refer to chapter 3.1 Data traffic Mirror and Capture.

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 3.6 Can’t hear other side voice in a call
Step 2: Check capture packet. you can filter keywords “sip || rtp”

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 3.6 Can’t hear other side voice in a call
You will find that the caller requests to send G.711 PCMA code（refer to point 1 of the screenshot）,
The called ONU will communicate with the G.711 PCMA code according to the requirements of the caller
ONU, but caller ONU sends G.711 PCMU code itself （refer to point 2 of the screenshot).
At this time, the two voice codes will be inconsistent and normal calls will not be possible.

Solution:

Contact your VoIP service provider and tell them your captured information
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 3.7 There was a lot of noise in a call
nphenomenon：When you are talking, you find that the called is talking intermittently or
there was a lot of noise in a call .

n Analysis:
This may be affected by the quality of the network, other services affect the transmission of
voice services. You need to set the priority of voice data transmission to the highest

Solution:

1. Replace the telephone line or telephone for test

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 3.7 There was a lot of noise in a call
2. Check whether the VoIP wan connection is normal

If Ping result is abnormal, you need to check whether the VoIP WAN connection configuration information
is correct. If Ping is normal, proceed to the next step

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 3.7 There was a lot of noise in a call
3. Configure the 802.1p value of voice to the highest
level

If you still have abnormal call after setting

the priority, Please refer to chapter 3.1 for
the method of capturing VoIP packets and send
it to VSOL technical support team for analysis.

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 3.8 VoIP Phone cannot talk normally
n phenomenon：This is a common problem in VOIP applications（topology is as follows）
At this time, The voice will not be heard by the VoIP phone on the ONU LAN side.

PSTN

OLT

ONU

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 3.8 VoIP Phone cannot talk normally
n Analysis：VOIP phone is in the intranet of ONU, It uses NAT technology for address
translation, When communicating with the external network, the signaling address has been
converted to the public network IP, but the media IP is not converted, it keep the private
IP of the Intranet ,As a result, the data from the external network cannot directly reach
the internal network address, so it caused this issue.

We can analyze this issue by capture packets

Step 1: Please refer to chapter 3.1 Data traffic Mirror and Capture.

Step 2: Check capture packet. you can filter keywords “sip”

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 3.8 VoIP Phone cannot talk normally

It can be seen from the interactive signaling that the ONU tells the SIP server the media address
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 3.8 VoIP Phone cannot talk normally

RTP send to Private network address, so the media data can't reach voip phone

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 3.8 VoIP Phone cannot talk normally
Solution:
enable ALG function on ONU web

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 3.8 VoIP Phone cannot talk normally
Both the caller and the called can hear each other enable_alg.pcapng

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 VoIP Trouble Shooting

The above troubleshooting is the analysis idea of VoIP related issues ，may be some issues that
are not among them.

If you encounter VoIP related issues , you can refer to this tutorial ,and refer to chapter 3.1
Data traffic Mirror and Capture. Analyze the capture or provide it to VSOL technical support
team.

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 Thank You

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