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Questions tagged [sip]

The Session Initiation Protocol, defined in RFC 3261, is an application layer signalling protocol for establishing and modifying long-running relationships between two or more peers. If including this tag, be sure to also tag your question more specifically to include the language you're programming in, as well as any specific libraries being used such as [pjsip], [python-sip], etc.

The Session Initiation Protocol, defined in RFC 3261, is an application layer signalling protocol for establishing and modifying long-running relationships between two or more peers.

More information at Wikipedia article on SIP

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Video call between eyebeam and baresip SIP clients

I am trying to achieve video call on 2 SIP clients Baresip Eyebeam Till now I have succeeded in getting audio stream both ways but the video stream is one way i.e iam getting the stream at the baresip terminal but I cannot see video at the…
user1490337
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Out Of Dialog NOTIFY message in SIP

It is possible to send Out Of Dialog NOTIFY message? I am working on a PBX and SIP phones are connected to it. I need to send the NOTFIY message to the phone on some change in PBX. I know that the phones must subscibe to the PBX and PBX can send…
Sunil
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asterisk load balancing using openser/opensips

I need to load balance incoming calls to asterisk. To do this, I have set up the Openser server in front of it and I loaded and configured the dispatcher modules to do so. What I want to do is that the Openser server will receive the calls and route…
Sara Ibn El Ahrache
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Is it possible to send a SIP notify message programmatically to a registered SIP device?

Is it possible to create and send a SIP packets programmatically to a registered SIP device ? I would like to send a SIP notify message, something like shown below: NOTIFY sip:alice@alice-phone.example.com SIP/2.0 To:…
joel
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What is going wrong in this SIP call? Multiple NOTIFY messages in a row before RTP established

the long string of NOTIFY messages happen after the called number answers. and after about 20-30 seconds the 503 happens and then the call connects fine with audio.
Adam Johns
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Wrong password issue in Samsung Galaxy SII

I am developing one SIP based application to make and receive a call.I have used shared preference to store the all the registration related data like user-name,password,context etc.This data are also stored in A2billing.And i have used Asterisk…
Juned
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VoIP, SIP, noice reduction, echo cancelation

I am tweaking an existing SIP solution for asterisk VoIP. It is working, but it has no sound quality control or no filters to improve the sound on bad lines. I searched google over and over for existing Delphi solutions in this area or some…
Runner
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RTP SSRC: How to know call direction

I am using jNetPcap to decode rtp from tcpdumps. Currently I use the SIP Invite Message and the source IP (and also checking via source ips..) to detect the directions (forward, reverse) from the call.. this is working but not really how it is…
Stefan
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Flash Media Gateway with SIP refer

I am trying to find out if the Flash Media Gateway (an add-on for Flash Media Server that supports Telephony integration) is capable of supporting the SIP REFER verb. In the current environment we have set up, whenever a SIP REFER message is…
javram
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android ngn stack and motorola XT912 android chashes

I am trying to use IMSDROID (android ngn stack) SIP stack on an android Motorola XT912. I've tested the android ngn stack on other android phones but there were no problems. Need some help. Thx!
just ME
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Syntax of H.264 SPS/PPS in SIP/SDP offer

According to RFC 6184: Annex B of H.264 defines an encapsulation process to transmit such NALUs over bytestream-oriented networks. In the scope of this memo, Annex B is not relevant. I see a lot of examplex, including in RFC6236, of…
Bob
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Sip settings without domain name

I want to develop a VOIP application for iPhone. I used the Sip library from Linphone but I can't register without a domain name. The server has no domain. Is there a way to allow the user to register with only the username, password and Server…
Alexandra Smau
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How To Create A Voice Chat application ( SIP Protocol )

I need to create a voice chat application . My preference is to use QT to develop client program as well as the server. The Clients are identified by a particular no. So when each client try to communicate with other client's , they should first get…
Akhil Thayyil
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SIP Demo without Registration

Im using the SIPDemo and it doesnt register with a Server. I can make calls but cant receive calls. I think the Problem is the registration. My Question: Is it possible to make a peer-to-peer call without a registration using the android API?
Johan
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Enabling debug logs on JAIN SIP (NIST implementation)

I'm developing a Java application based on JAIN SIP with the NIST implementation and would like to enable/view SIP stack debugging. I can't find a working way to achieve that - any help would be appreciated. Thanks! Joe
Johannes Liebermann
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