Hi experts, I am new in android. My query is regarding SIP audio call and its call flow. After SIP connection establishment, app gets onCallEstablished and then it calls call.startAudio(); After that I suppose that the connection is established between mic and rtp sockets/ports and all audio packets coming from mic are redirected to the peer SIP users using rtp protocol. 1. So, what are the function calls after startAudio() and modules involved till it makes connection till mic audio packets. 2. On the reverse side flow i.e. After the connection is made, what is the flow from mic audio packets till rtp socket and port and how it is redirected to SIP peer user. What are the flows and modules involved.
Please help me in whatever little you can provide. Anything can help !!! Thanks you very much in advance. -Nagendra -- You received this message because you are subscribed to the Google Groups "Android Discuss" group. To unsubscribe from this group and stop receiving emails from it, send an email to [email protected]. To post to this group, send email to [email protected]. Visit this group at https://groups.google.com/group/android-discuss. To view this discussion on the web visit https://groups.google.com/d/msgid/android-discuss/c20ab7bd-aba2-4129-a0b0-bde3a295a00a%40googlegroups.com. For more options, visit https://groups.google.com/d/optout.
