SIP over HSPA/EDGE: how to overcome packet loss when establishing a connection (the first couple of seconds)?

Question

Asterisk

Good day!
I am new to VOIP and asterisk in particular.
Out of personal interest and in order to save money on communications, I set up an Asterisk server (with FreePBX) with several SIP trunks and a modem. Everything works great until it comes to mobile internet.
Built-in SIP clients of Nokia smartphones are used as clients.
Empirically, it was found that for networks with high losses and variable jitter, the iLBC codec is best suited.
At the moment, calls via EDGE work fine.
But as always, there is one thing: the first couple of seconds after the channel is switched to the “answer” state (when the transmission of RTP packets begins), the interlocutor of the person sitting through the mobile network hears gurgling. It takes a couple of seconds, the connection settles down and you can talk perfectly.
This moment is annoying. How to win?
PS: TSL authentication and SRTP encryption voices are used (checked, it does not affect).
PPS: if anyone is interested in how to set up TLS + SRTP on Asterisk + Nokia, let me know - I will write an article.
upd: I checked again, the problem is pronounced on the HSPA connection (operator 3mob, Ukraine). On EDGE, the first second is "eaten up".
There is a suspicion that this is somehow connected with 3g timers.

Answer 1

Try (in AEL example):

answer;
wait(2);