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SIP ptime

 ptime is packetization timer 

Auto-Answer

 https://voip-sip-sdk.com/p_7354-how-to-implement-auto-answer-using-csharp.html

SIPP

 https://tomeko.net/other/sipp/sipp_cheatsheet.php?lang=en working commands for sipp sipp -sf uas.xml -i 172.18.1.41 -p 5060 sipp 172.18.1.41 -sf uac.xml Regiteration :- https://openplatform.xyz/sipp_register_uas_and_uac.html sipp -i 172.18.1.41 -p 5060 -sf register_user_auth.xml -inf users.csv -infindex users.csv 0  sipp 172.18.1.41 -sf register_uac.xml -inf registar.csv      FreeSWITCH to SIPP Client communication commands    sipp -sf uac.xml -s 1003  172.18.1.221:5080 -trace_msg -l 15 -d 60000    Call Hold scenario  ------------  sipp -sf Hold-UAS/uac-hold.xml -s 1003  172.18.1.221:5080 -trace_msg -l 15 -d 60000 -m 1    reinvite from SIPP to freeSWITCH  --------------------------------    sudo sipp -sf Hold-UAC/uac-hold.xml -i 172.18.1.221 -s 1003 -m 1 172.18.1.221:5080 -trace_msg -l 15 -d 60000  

List of Open Source SIP Solutions

 https://telecom.altanai.com/category/session-initiation-protocol-sip-frameworks/

List of SIP RFCs

 https://www.dialogic.com/webhelp/BorderNet2020/2.2.0/WebHelp/sip_ov_2020IMG.htm

Call Hold

 https://sipsorcery-org.github.io/sipsorcery/articles/callholdtransfer.html  

File transfer to board

 Connect the lan cable from ONT to Laptop Download Solarwinds TFTP Server Install the TFTP Server Goto File-> Configure (Check the TFTP Root Server directory as C:\TFTP-Root) Keep the voip_kernel.ko(file) in C:\TFTP-Root check the ipconfig in cmd (you will get 192.168.1.254 as Gateway IP and System IP as 192.168.1.64) 192.168.1.64 will acts a TFTP Server and your board will act as TFTP Client ssh to 192.168.1.254 using Putty Now do the tftp command in .254 putty window  tftp -gr voip_kernel.ko 192.168.1.64 Now check the MD5SUM on your windows and linux system both should match C:\TFTP-Root>certutil -hashfile voip_kernel.ko MD5 MD5 hash of voip_kernel.ko: 88e8f9781b246b4e63429f597e21c280 CertUtil: -hashfile command completed successfully. root@Cortina:/home/ONTUSER# md5sum voip_kernel.ko 88e8f9781b246b4e63429f597e21c280  voip_kernel.ko Both the MD5SUM macthes means both are identical files Original file md5sum of original file  ------ root@Cortina:/home/AdminGPO...

ONT Board bring up

  Connect the lan cable from ONT to your laptop Goto "View Network Connections" Now goto "Ethernet connection" -> "Status" -> "Details" In this you check IPv4 Gateway IP(I got 192.168.1.254) and set mask as 255.255.0.0 Open 192.168.1.254 in browser goto Software Upgrade option Download the image from path http://135.251.205.50/IMAGE/BBDR2104OpenWrtB/ BBD_2104H.508.tar download it Now extract this tar you will get multiple builds inside check AONT SW Strategy page for checking which built you need for hardware Now check http://fn-cm.sbell.com.cn/wiki/index.php/AONT:_SW_image_Strategy#SW_image_name_Strategy (for your device image) Now upload your image(3FE49544IJIL08) in "Software upgrade" option goto 192.168.1.254 username: AdminGPON password: ALC#FGU for shell password generation(goto https://dhbusec2.china.nsn-net.net/DHBUsec/ckp_handling.php?ckp=home&saml_ru=1 -> CLI 2nd level Password to enter Shell (ALU02335783)) Cortin...

Power related parameters

  Measuring VoIP Quality with SIP and RTP (catchpoint.com) rx gain, tx gain, slic gain companding DCProfile 1. A SIP-channel only patch that works for alaw/ulaw (a bit of a kludge) 2. A technology independent patch that works for all codecs[1] that adds a SetGains(txgain,rxgain) function for adjusting incoming channels, and adds a V(txgain:rxgain) flag to Dial() for outgoing channels (much less of a kludge). Range extenders - nordicsemi.com

Call Conference

 https://www.tech-invite.com/fo-sip/tinv-fo-sip-service-11.html https://www.dialogic.com/webhelp/img1010/10.5.3/webhelp/description/sip/sip_3pcc_call_flows.htm

SIP when interacting with Network , issue is on network

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 Issue is  Why the G240W-B ONT is not registered in the Sip server? Not able to see REGISTER message itself on SIP traces Solution: Issue can be in DHCP, DNS server, IPv4/IPv6 or PPPoE server Try to check with traces for PPPoE in this case

Terminology

 https://en.wikipedia.org/wiki/Echo_suppression_and_cancellation https://wiki.analog.com/resources/tools-software/sigmastudio/toolbox/adialgorithms Telecordia Standards like    GR-283-CORE pulse timer - https://www.wisdomjobs.com/e-university/programmable-logic-controllers-tutorial-523/pulse-timers-14725.html#:~:text=Pulse%20timers%20are%20used%20to,output%20from%20some%20initiating%20input.&text=When%20there%20is%20an%20input%20to%20In%201%2C%20there%20is,elapsed%2Cthe%20timer%20contacts%20open. line feeding -  feeding of line changes from standby to active to different state depends on on-hook off-hook , and other line status howler tone - howler mode -   ring cadence -  polarity reversal -  Polarity of the line will be reversed,  +- of a/b wire of RJ11 polarity means  dc voltage,  it is  the directional flow of electrons from one pole to the other https://www.ixysic.com/home/pdfs.nsf/www/an-118.pdf/$file/an-118.pd...

Types of modems and codecs

V.92 modem --> US Robotics  V.91 modem Bell 103 --> ITU V.21 --> 300 buad rate Bell 212a --> ITU V.22 --> 1200 buad rate ITU-T Rec. V.150.1 (01/2003) Modem-over-IP networks http://what-when-how.com/voip/fax-and-modem-tones-basics-voip/ H.248 PCM AVP AMR-WB/AMR

Types of hooks & ringtones

 Flash Hook OnHook Offhook types of tones ---------------- Dial tones Ringtones Ringback tone Congestion tone MWT (Message waiting tone) Howler tone SpecialDailTone StuttleDailTone Call waiting Tone https://en.wikipedia.org/wiki/Call-progress_tone

SIP companies

  Impiger Technologies Pvt. Ltd CISCO Juniper

DigitMap

https://sipfoundry.atlassian.net/wiki/spaces/sipXecs/pages/491748/Digit+Maps+used+to+Define+the+Dial+Plan  https://www.symetrix.co/wp-content/uploads/2016/04/2014-07-VoIP-Digit-Map.pdf https://www.poly.com/ca/fr/support/knowledge-base/kb-article-page?lang=en_US&urlName=knova-33287-how-to-configure-understand-digit-map-on-polycom-phones https://support.polycom.com/content/dam/polycom-support/products/voice/soundstructure/other-documents/en/understanding-digit-maps-tech-tip.pdf https://knowledgebase-iframe.polycom.com/kb/viewContent.do;jsessionid=EBE29064D5174273CE8E30453DAB9FF4?externalId=38743 https://documents.polycom.com/bundle/obi-ag-640/page/OBiPhoneAdminGuide/call_controls/TOC_Digit_Map_Configuration.htm T Wait for timeout when dialing "R" Digit map extension letter R indicates that certain matched strings are replaced. For example, R555R604R would replace 555 with 604. "E" Specify ending termination digit which follows (usually * or #) The letter "P...

State Machine Issue

 In SIP INFO we send the extra digits in "subsequentDigit:" section but in our case digits were missing inside SIP INFO SIP INFO is used to transmitt the dialed digits. As per logs we were able to observe that digit got missing because our state machine was in CC_WAIT_FOR_PLAT_RESP instead of  CC_SECOND_DIAL because MSG_STOP_SIGNAL_REQ is triggered but  MSG_STOP_SIGNAL_RESP is not receiving hence it's stuck in CC_WAIT_FOR_PLAT_RESP . [2021-12-03 18:22:01:958][WARN]msg: MSG_EVT_DETECTED_IND [2021-12-03 18:22:01:958][DEBUG][pal_msghandler, 3239] pal_fn_voipManagerMsgHandler: MSG_EVT_DETECTED_IND Received FOR TERMINAL:1, item_id=2002 [2021-12-03 18:22:01:958][DEBUG][pal_msghandler, 5117] pal_fn_voipManagerMsgHandler: MG_E_Digit Received:2 [2021-12-03 18:22:01:959][DEBUG][pal_fsm, 19173] CC FSM:CC_WAIT_FOR_PLAT_RESP: Event Pal_en_digitReceived received for CallId:5 on Terminal:1 dNextState=35 [2021-12-03 18:22:01:959][DEBUG][pal_msghandler, 6964] Exit pal_fn_voipMan...

SIP VoiP Debugging

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SIP Port range is 5060-5080 And RTP Port range is 16384-32768 Kindly change the port in xml as 5060 is conflicting with 5000-5100 port range <qos-dscp>0</qos-dscp>     <local-ports>5000-5100</local-ports> To    <qos-dscp>0</qos-dscp>     <local-ports> 16384-16484 </local-ports> https://supportdesk.win911.com/support/solutions/articles/24000038761-port-ranges-for-supported-sip-and-voip-providers https://wiki.innovaphone.com/index.php?title=Howto:What_Ports_are_used_for_Signaling_and_Voice_Traffic_in_SIP_and_H.323%3F https://freeswitch.org/confluence/display/FREESWITCH/QoS   http://startrinity.com/VoIP/VoipTroubleshootingBook/VoipTroubleshootingBook.aspx https://www.voiceelements.com/docs/programmable-voice/faq/why-am-i-missing-digits/ https://blog.kolmisoft.com/post-dial-delay-pdd/ https://andrewjprokop.wordpress.com/2013/09/27/dtmf-and-rfc-2833-4733/ https://www.nojitter.com/...