Top "Sip" questions

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.

How to do peer-to-peer communication in an iPhone app?

I'm trying to write a simple chat application for the iPhone (as an experiment). Is there a simple way for …

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ast_sockaddr_resolve: getaddrinfo(): Name or service not known

I have got this error: ast_sockaddr_resolve: Getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known …

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How to provision a test user in kamailio?

I have just (for the first time) compiled and installed kamailio, following this guide. For configuration, I am following the …

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Is sprop-parameter-sets or profile-level-id the SDP parameter required to decode H264?

I'm trying to understand what is the required parameter in SDP to be able to decode H264 from RTP packets. …

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VoIP Integration in App & Web

I have a very general question on how to implement VoIP for our current mobile & Web App. (we have …

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Linphone for android is not working/missing libraries

I am trying to run linphone code which I get from git://git.linphone.org/linphone-android.git --recursive . After downloading …

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Android SIP with CSipSimple and pjsip

I am writing a custom Android application that allows the user to make VOIP calls using SIP. When the user …

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Integrating Asterisk with WebRTC - ground up

I am trying to integrate Asterisk with webRTC. There was a query posted here but it barely provides any solution. …

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RTP Proxying with OpenSIPS/Kamailio/OpenSER

I have an OpenSIPS server which listens on multiple IPs. If I route the calls back out to the next …

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How to tunnel voip traffic in blocked networks

I have an Android VOIP application. As some networks block VOIP traffic I wold like to find some way to …

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