clone twinkle. Sip softphone web based
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Hi, i'm developing a simple client "web based" in python and i use a labrary of twisted that allows me to call two phone numbers and bridge them.
That is a click-to-dial service.
i need an help for solving a problem.
I use asterisk as proxy server.
What i do is this:
1) Call from my client number A towards asterisk (so i have one sip invite from my client to asterisk and one sip invite from asterisk to phone number). When i send this INVITE i specify in the sip message body some codecs to be used,my destination ip and port (where i expect to receive rtp packets). In the same time I open an rtp listener and a rtcp listener.
2) When user picks up the phone A (i receive a sip message OK and then i send an ACK), I send a REFER to asterisk with header Refer-to: number B.
3) Asterisk allows REFER method and so it answers my client with a 202 accepted.
4) While sending my client some NOTIFY messages, asterisk invites number B.
5) When user at B picks up the phone, my client sends a BYE to asterisk. In the same time, asterisk bridges the two channels ( number A and number B)
6) Now the A and B can talk each other.
Seeing rtp debug on asterisk, I see that there are no rtp packets sent between two numbers, and so there is no voice in the call.
But if I do the same thing with a softphone named twinkle, i see in asterisk's rtp debug a lot of rtp packet sent from A and B and viceversa.
I guess it is because i don't manage rtp packet sent to my client.
Have I to manage rtp packet or it is not necessary?
p.s. all my peers in [url removed, login to view] have reinvite=no
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