sip

sip

Is this the valid contact header and can you explain the parameters

How to get audio stream properties (format and sample rate) of my VoIP provider?

SIPp call between two registered UACs

Ringcentral-web-phone (sip.js) - Subscriber not found

RingCentral-Call-JS - 603 - Too many contacts

OpenSips Remove Header Value

SIP: UAC UAC role change due to re-invite

CSeq INVITE sip request 1 and 2

Freeswitch outbound calls dropping after about 30 seconds due to ACK timeout [closed]

j.sip.TransactionAlreadyExistsException

Must the "telephone-event" have the same frequency as the codec used in the call?

How does SIP/RTP determine two endpoints are on the same LAN?

How to make a SOS/emergency call using pjsip

SIPp returns error like "Ignoring non IP{4,6} packet!"

SIPREC Protocol with FREESWITCH

Can one REGISTER request contain multiple To Header Fields and what would be the scenarios?

Showing errors in kamailio syslogs for Register messages

SIP dialog established via UDP - what is the transport AFTER a large SIP request has been sent via TCP?

How to specify tshark script what interface to capture on [closed]

How to get Status code for Trying and Ringing in JsSip

SIP 200 OK to INVITE followed by 100 Trying

SIP/2.0 439 First Hop Lacks Outbound Support error

IMS - Forwarding calls flow in SIP - SDP negotiation

[SIP]Response on Receiving 488 (Not Acceptable Here)

kamailio - modify URI of incoming INVITE

kamailio: how to send replies with port number in VIA header

Is the following sintax rfc 3261 compatible?

SIP server forwarding

exosip always answer twice when receive invite event

pjsip (pjsua) notification when remote user answers the call

Does sip provider can control sip video quality?

SIP auth with ejabberd

Process INVITES when recording is enabled on both parties phones in Cisco CUCM

Is a sip address the same as an email address?

Why Getting SIP/2.0 603 Decline TV1 or SIP/2.0 480 Temporarily Unavailable when make outbound call from SIP and destination app is terminated

SIP over TLS bandwidth consumption

How to get Kamailio to set `Record-Route` header to internal IP for internal leg of call?

How to send WebRTC audio from Kurento to Avaya phones

Freeswitch wrong caller id number after bridge

Freeswitch sending empty RTP in time with receiving RTCP

Which PJSIP library level to use for B2BUA

PSTN Call is not working over opensip instalaltion

Pjsua (pjsip client) does not want use TCP

FreeSwitch: Send image data on SIP body / multipart on call

Pjsip/Pjsua video problem : frame buffer too small

Is there a way to adjust H264 profile level in the Ozeki VoIP SIP SDK?

Linphone opus codec sampling rate

Is user part in sip 'Contact' header madatory?

PJSUA: verbosity of debug information on the application console

VoIP delta spikes below 20ms, causing the jitter to change

Two instances of microSIP on the same Windows7 [closed]

VOIP/SIP Call Hold related questions

Is it possible when 200OK for UPDATE request coming after 200OK for initial INVITE in SIP?

Problem with sending Invite in Linphone 4.x

is SIP:480 after 180 ringing same as SIP:480 after 183?

Opensips 2.4 can't make outbound call

Weird network issue

What's wrong with my VoIP setup on Edgerouter X (Telekom Germany)

python3-pjsip(pjproject2.9 pjsua) custom sdp

Is there a way to retrieve the sessions allocated by rtpproxy?

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