sip

sip

How to get media attribute from SDP body in pjsua2?

How to get video streams from FusionPBX video conference room

How do I ensure that BYE messages bypass a SIP proxy?

Kamailio - Dispatcher determine availability via http?

SIP registration process

Is BYE hop by hop? or end to end?

Kazoo/Freeswitch bridged call from restricted number, how to set sip privacy off

osip_message_to_str fails on a message I created using the message/parser functions

SIP Content-Type charset, in which standard is it defined

Is there any RFC standard for "Raise hand" functionality in SIP calls?

Free SIP tool support header normalization

How to use caller ID instead of "s" in SIP refer method?

how to record the voip call using sipsorcery sdk?

How do you do NAT traversal for RTP media using Kamailio for signalling?

kamailio Privacy header and CANCEL

Why doesn't a SIP server require authorization for this INVITE?

SIP UPDATE Media Description with 0 fmt

What are the possible values of “prot” parameter of “SecurityClient” header in a SIP Request?

configure//make error on Ubuntu Pjsip PJproject

Restcomm USSD - do not return a dialog

Incorret fmpt of telephone-event in received INVITE

How to get value of SIP header in Freeswitch?

How is a SUBSCRIBE request can terminate a SIP dialog?

H323 Video with Freeswitch mod_h323 possible?

Must SIP UA wait for Ack before re-Invite?

Why can't we use "Local tag" & "Remote tag" to identify a dialogue?

Which takes precedence VIA or Contact header?

Type of data in a trace of type sip

Linphone SIP client not working Behind NAT

Duplicate payload type for same codec but different fmtp line is it valid scenario?

I have a PCAP file with SIP/RTP... can I replay it into a switch to test compatibility?

sipcmd doesn't detect remote connection state change

Cannot create webrtc call using sip.js

How to allow or passthrough a (re)invite in Freeswitch dialplan

Sip UPDATE method

E2U+sip vs sip+E2U

DTMF digit with SIPP test

Behavior of sip INVITE

Freeswitch Dial-String for SIPs + SRTP outbound call

Multicast call hangs up after 30 seconds

SIP Route parameters: r2 and rpp

There is a voice delay while using linphone-sdk

Sip: Prevent INVITE request information spoofing

SIP Re-invite and user=phone

How can do accounting manually in opensips

SDP Offer/Answer model with DTMF rtpmap/fmtp mismatch

Cannot make calls from Skype for business mobile clients to Cisco

Stop SIP SUBSCRIBEs from endpoint

Plivo Inbound Sip trunk - issue with destination number

Switch wake on LAN

openSIPS setup an onreply route if the call is picked up

SIP CSeq for INFO and INVITE methods

Skype-in Telephone Numbers no longer accept Toll Free SIP "From" or P-Asserted-Identity

How to use allow_contact_rewrite in pjsip in android

Using SDP media type application with RTP/AVP (m=application <Port #> RTP/AVP <Payload>)

Voice call from app to phone using sip and gateway

IMS Clearwater configuring Application Server

Unable to establish calls with Freeswitch sometimes

Tomcat 7.0.82 Release for Restcomm Sip Servlets

enterprise originate ignores SIP 302

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