PJSIP signaling

Must have already completed large PJSIP projects. Do apply if you do not have experience in PJSIP and SIP.

Must have strong understanding of SIP and PJSIP. Job will require you so show sample of PJSIP work.

References on PJSIP projects. Payment will be delivered only have a demo of the delivered code. Screen share will be fine.

demo code will need to have wire shark traces. When bidding on the job please explain your pass experience in PJSIP.

Events

• Ring

o Ring(int ch)

• Incoming call

o Incoming (int ch, string sCaller, string sCallee, string sDestAddr, string sViaAddr, string sFromIP, ushort nFromPort)

• Connect

o Connected(int ch)

• Disconnect

o Disconnect(int ch, int code, string desc)

• Session Timer

o On_Timer(int ch)

Functions

• MakeCall

o ChannelNumber,From,To,UserName,Password)

• Answer Call(channel)

• Disconnect/Cancel

o 302 must not redirect but just disconnect

o DisconnectCall(channel)

• SendRing(channel)

• GetCodec available

• GetCodec in use

• Current Status of a channel

• Get Next Free Channel Number

• StartTimer(channel)/StopTimer(channel)

• Join Two calls

 Sample (channel1,channel2,bool UseRTP)

Settings

• Listen IP[]

• Port[]

• Number of threads

• Max ports

o Return 503

• Max cps

• codec

• Events need to be in its own thread.

• Check for deadlocks. I want PJSIP to use over 10 threads or as many as I can let it have.

• CPS Required 1k+ (understanding it’s up to the hardware)

• Ports Required 10K

• C++ wrapper in 64 bit so C# can use it.

• Basic sample in c# to use c# wrapper

• Need c# sample program tested for long to make sure no memory leaks

• Need option to do RTP and NO RTP. RTP in PJSIP Bridge I was told had bad performance. Need to make sure this does not kill the program.

When doing a join using Audio I need a way to join to Trans code RTP or not to.

1 year support on project

Programación en C++ VoIP

Nº del proyecto: #5083898

Sobre el proyecto