clone twinkle. Sip softphone web based

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 [login to view URL] have reinvite=no Before bidding : PAYMENTS 1. 20% for starting job a. 80% ON COMPLETION 3. Money sent via PayPal Work rule - you will speak via chat or skype with me - if you can't solve the problem i want a refund of deposit Before bidding - Don't bid if you are not sure for solve this problem. if you have duobt please post me a message.