Questions tagged [sip-server]

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.

The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.

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

I am facing an issue regrading the Sip call my flow is as below. A Party sends INVITE to SIP agent SIP Agent sends INVITE to B party B replies with 100 (Trying) B sends SIP 488 (Not Acceptable Here) Cal Disconected Sip Agent sends 488 (Not…
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Converting SIP2(Standard Interchange Protocol) Request to Web API Request and web API Response to SIP2 Response

We have a requirement from one of our clients. They had in-built library management system(LMS) in .net c# and have API's to interact with LMS. Recently they introduced KIOSK and RFID system in library for issue, submit etc. of books through KIOSK…
Snziv Gupta
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Setup SipSorcery Server

I am new in SIP Programming, while searching on Google "voip sdk c# open source github", I got reference of sipsorcery. I had downloaded it and try to play with samples but I am unable to execute it. I had search tutorials on google and youtube but…
Dave Kapildev
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Portsip SDK on Android 10. Registration or Refresh does not get back response

This is my login function: fun login( userName: String, password: String, serverHost: String?, serverPort: Int, transportType: Int, displayName: String, authName: String, userDomain: String, srtpPolicy: Int, localP: Int?,…
rosu alin
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Is there a way to prevent sending multiple SIP messages in a single TCP packet?

I am testing the integration of a SIP Server(just a PBX) with another PBX. The incoming call comes into the PBX and is sent to my SIP Server. Once the call arrives on my SIP Server, it is then again sent back to a user who is registered to my PBX.…
Bashab
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How to configure officesip server audio codec to be compatible with jssip

I am getting the following error message while trying to make call with jssip to officesip server Incompatible SDP SIP/2.0 488 Not AcceptableHere I found out that this is due to the fact that no common codecs between jssip on browser(Mozilla) and…
j4rey
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How do I ensure that BYE messages bypass a SIP proxy?

I am trying to build a stateless SIP proxy similar to the one in this diagram: It behaves as it should when "Alice" hangs up first, the BYE message goes directly to "Bob" and does not touch the proxy. However, when "Bob" hangs up first, the BYE…
Adam V.
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SIP library for watchOS 5 development kit

In my case, I need to make an in-house communication system using Apple watches ( watch to watch, using WiFi only). There I'm planning to use SIP protocol for voice communication *And already found the SIP Library linphone-swift-demo And it is not…
HomiFox
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linphone use SIP message for sending message in chat room

I am building an android application where I am using LinPhone Lib for audio/ video and sending a message using VoIP. I read a lot of blogs and finally, I got an EasyLinphone GitHub Code that is using LinPhone Lib for audio/ Video calling. I read…
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How to detect android.permission.USE_SIP

Now I have a serious problem with my manifest. I don't know why even after had android.permission.USE_SIP in manifest, this is not recognized. I use this code to detect. if (ContextCompat.checkSelfPermission(this, Manifest.permission.USE_SIP) …
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Cannot create webrtc call using sip.js

I am not able to create a Webrtc call using sip.js on FreeSWITCH. I believe there are some issues with Freeswitch configuration, but I'm not able to figure out where the issue is and how to figure it out. Any help would be highly appreciated. I've…
ambach 66
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Sip UPDATE method

I am new to sip protocol.I understand the normal sip mechanism like how it works.I know about sip re-invite method which is useful to update the SDP(Session Description Protocol) parameters.But recently i found UPDATE sip method which also do the…
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need example of PJSIP for android

I want to connect to SIP server using PJSIP through android, how to integrate in a android studio. I have taken example of csipsimple , but i am unable to connect the SIP server using PJSIP.
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Installing Freeswitch on Google Cloud VM

I have installed Freeswitch (centos7) on a Google Cloud VM. This is just a test instance. Haven't made any changes to the config yet. The server starts up fine but I can't get any audio through while testing. Have opened the firewall as per the…
vjm
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Call mobile from computer with SIP

Someone knows some app what can call to mobile phone like a webphone.... I found jssip or sip2sip but i can't put into operation. This is for a telemarketers company
nickokss
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