SIP VoiP Debugging

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/working-dtmf-transmission-packet-level

https://www.dialogic.com/webhelp/NaturalAccess/Release9.0/Fusion_VoIP_API_Dev_Manual/Transferring_DTMF_digits_according_to_RFC_2833.htm

https://www.3cx.com/blog/voip-howto/sip-invite-header-fields/

https://www.voipmechanic.com/dtmf-issues.htm

https://knowledgebase.simplicityvoip.net/knowledge/troubleshooting-call-failure-specifics

https://www.nextiva.com/blog/voip-problems.html

https://wiki.freepbx.org/display/PC/DTMF+Issues

https://www.audiocodes.com/media/14274/audiocodes-quick-reference-guide-voice-quality-troubleshooting-for-session-border-controllers-sbcs-and-gateways.pdf

https://support.exotel.com/support/solutions/articles/3000091334-troubleshooting-with-dtmf-gather-ivr-

https://cdn.ttgtmedia.com/searchVoIP/downloads/Building_a_VoIP_Network_Ch[1]._8.pdf

Call Hold Tone(DTMF) played for 60 seconds only not 89 seconds when call hold went to call resume

Code fix:-

bool LocalHoldToneGeneration::playLHTone(Pal_ty_u16bit txn_id, mg_term_id_t scn_term_id, Pal_st_error* pErr)

{
Pal_ty_retVal ret = pal_fn_sendStartSignalBasic(txn_id, scn_term_id, LOCAL_HOLD_TONE, MG_TIMEOUT, 60000, pErr); // 60 seconds hardcoded changed to 1000 secs

https://thanhloi2603.wordpress.com/2017/06/03/understanding-sip-call-onhold/

https://voipdocs.io/en/articles/309-understanding-sip-packets

https://www.softarmor.com/wgdb/docs/draft-ietf-sipping-mwi-00.html

https://voip-sip-sdk.com/p_7153-rfc-3265-sip-event-notification.html

https://voip-sip-sdk.com/p_7172-rfc-3842-message-waiting-indication.html

https://docs.skyswitch.com/en/articles/676-blf-and-mwi

https://saevolgo.blogspot.com/2012/07/asterisk-behind-kamailio-voicemail-mwi.html

https://startrinity.com/VoIP/Resources/sip-introduction.pdf

I have checked in that no CLIP parameter can be used to trigger MWI

As per specs https://www.etsi.org/deliver/etsi_ts/183000_183099/183043/01.02.01_60/ts_183043v010201p.pdf
CLIP is triggered during INVITE message(Call Setup Message) :- check section D.2
MWI is triggered during NOTIFY message:- :- check section D.3

NOTIFY message is triggering FSKIndication in our code as given below:-
void pal_fn_process_notify_info_rcv_before_reg200(Pal_ty_u32bit   dLineId)

{

     pSubscribeData->setRcv_nty_bf_reg200(Pal_en_false);

        if (pSubscribeData->getmwi_NotifyKeepInfo() == Pal_en_true)

        {

            pSubscribeData->setmwi_NotifyKeepInfo(Pal_en_false);

            if (IDLE_ONHOOK == pGlbTermProfile[dLineId].dTerminalState.dCurrentState)

            {

                if (pSubscribeData->getmwi_waiting_NotifyKeepInfo() == Pal_en_true)

                {

                    PAL_REGSUB::Subscribe_Handler::Instance()->DoSendMsgToTerm(dLineId, Pal_en_fskIndication, Pal_en_true);

                }

                else

                {

                    PAL_REGSUB::Subscribe_Handler::Instance()->DoSendMsgToTerm(dLineId, Pal_en_fskIndication, Pal_en_false);

                }

            }

        }

Our device driver should provide the parameter & feature to above layer

Generic MWI parameters available to both BCM & MTK:-
    <message_waiting_mode>dial-tone visual</message_waiting_mode>

    <vmwi_refresh_interval>0</vmwi_refresh_interval>

                      <MWI>

                             <Event>

                                    <UaProfile>Disable</UaProfile>

              <MessageSummary>

                      <apply>Yes</apply>

                      <SubscriberUri>null</SubscriberUri>

                                                <MessageWaitingIndicatorTone>

                                                        <apply>Yes</apply>

                                                </MessageWaitingIndicatorTone>

                                                <VisualIndicator>

                                                        <apply>Yes</apply>

                                                        <RefreshTimer>1800</RefreshTimer>

                                                </VisualIndicator>

              </MessageSummary>

                             </Event>

                      </MWI>

FSK can be triggered in case of multiple services such as SMS, CLIP or MWI services these all are different sets of feature
https://yadda.icm.edu.pl/yadda/element/bwmeta1.element.baztech-article-BPS2-0027-0052/c/httpwww_itl_waw_plczasopismajtit2004266.pdf


MWI and CLIP both are different features

VMWI is used for Voicemail (https://en.wikipedia.org/wiki/Message-waiting_indicator)

CLIP is used for Caller ID display (https://en.wikipedia.org/wiki/Caller_ID)

Both are different set of features, but VMWI can be used for Caller ID feature also but opposite cannot happen.


DHCP SIP Interaction

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/cube-book_chapter_01001000.html

https://www.psn-web.net/cs-im/Global/SIPPHONE/sipphone_net/technic/TGP7/02-01-01_Pre-Provisioning&Provisioning%20flow_Rev.1.1.pdf