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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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How can I view output of tshark -V via Wireshark or similar?

Recently updated my Wireshark on a server, and lost the ability to use -R and -w from the CLI. Since I'm tracing SIP and RTP calls, I need to use -R and not -f. I found out using -V is very useful (shows the packet tree on screen) and then I can…
Muhammad El-Sergani
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Error while using a g729 codec in SipDroid

I am developing a SIP application to make and receive a calls. And i want that application to support g729 codec. First i have tried with SipDroid an open source project and i have followed the this steps (followed the comment from 149 to 160) to…
Juned
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How to share an actie sip session between activities? (Android)

My app connects with a sip server in a couple of activities. I haven't found a way to pass a registered sip session to the next activity. This means that every time the user goes to a new activity inside the app, the app needs to "reregister" at the…
lswartsenburg
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Android SIP Client

Which is the better way to start the SIP client for android using any external libraries.Since the native inbuilt SIP libraries work only with versions greater than 2.3.1.Looking for the better and easier solution to build the SIP Client for lower…
user961524
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Jain Sip processRequest Method in SipListener is not called

i got a strange Problem using Jain Sip (dowload today Version Jain-sip-ri-1.2.2014.jar). I'm connecting to a PBX to do some uaCSTA stuff with it. The Connection works fine i also can execute MakeCall Commands etc. but after some time (~20Minutes)…
outofBounds
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SIP outgoing call is not ended properly

In my application when i am making a outgoing call and if i ended it from my application its ended properly but when call receiver ends that call, sometimes it ends sometimes its not. i debug the application many times and came to know that…
Juned
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Understanding Asterisk Server features

I need to ask few question about Asterisk 1) Does ACL mean by Access Control list here ?If yes than how could i use it? >ip show user 6001 * Name : 6001 Secret : MD5Secret : Context : DLPN_Admin Language…
Arham Ali Qureshi
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Sip authentication --response value

I'm trying to authenticate myself with my asterisk server but I don't think that I am formatting the authentication header correctly, specifically the response value. I'm using the following formula to create the MD5 hash for the value: …
tier1
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SIP service to be used for incoming call

I am trying to automate incoming call to an Android Phone. I have a sip client on my PC which supports call to Phone Number. For this I need a SIP account which will allow me to make a call to Actual Phone Number. This SIP service should be…
Shrikanth Kalluraya
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Does Android 4.0.3 support SIP?

If yes then why SipManager.isApiSupported(), SipManager.isVoipSupported() returns false? Is there anything to be done to make it Sip supported? I have tried those codes in the emulator.
Juned
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SIP: how to know, for which REFER the NOTIFY came?

Suppose, we have a following dialog: Agent A Agent B | | | F1 REFER | |----------------------->| | F2 202 Accepted | |<-----------------------| | F3 NOTIFY | …
weekens
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.NET Voip integration

I have this requirement: A customer presses a button on a mobile phone. This sends a web service request to the server to initiate a call on customer's behalf. The server makes a phone call, which plays a pre-recorded message from the customer, to a…
user1044169
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how to get incoming call alert dialog on main or home screen in SIP application?

I am developing the SIP application for call and i have used AlertDialog for incoming call notification but its not feasible because when application is running on the background and someone calls, the "call screen" isn't been launched.what should i…
Juned
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Softphone client to use with g729

Is there a softphone ( opensource, freeware, or trial version) with G.729 codec ? I just need to make a call and see how it goesthrough a sip protocol on Asteriks. Thank you on advance.
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how to invite minimum 2 persons to a conference room using asterisk and call files

I am using asterisk 1.8 on ubuntu 9.10. I want to ask if it's possible to invite more than 2 sip clients to a conference room using call files? A sip client wants to create/ or has a conference room and would like to invite other 3 persons. how…
user1222905
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