Задержка перед вызовом и несколько invite . Второй вопрос

В продолжение http://asterisk-support.ru/question/66233/zaderzhka-pered-vyzovom-i-neskolko-invite/

Посмотрел sip set debug on. Звонок идет с 3002 на 3001. Приходит первый invite без авторизации, astersisk сразу отвечает на него 401 Unauthorized, телефон присылает invite с авторизацией, и тут происходит задержка: приходит invite, asterisk начинает его обрабатывать (если я правильно понимаю):

 
<--- SIP read from UDP:192.168.6.147:5060 --->
INVITE sip:3002@192.168.1.129;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.6.147:5060;branch=z9hG4bKf3ae399d6c8efe4097e73c7d70d1c104;rport
From: "3001" <sip:3001@192.168.1.129>;tag=2038128166
To: <sip:3002@192.168.1.129;user=phone>
Call-ID: 2522539599@192_168_6_147
CSeq: 3 INVITE
Contact: <sip:3001@192.168.6.147:5060>
Authorization: Digest username="3001", realm="asterisk", algorithm=MD5, uri="sip:3002@192.168.1.129;user=phone", nonce="21b27acb", response="d77617128c8b73645a711fd72c84c131"
Max-Forwards: 70
User-Agent: C530A IP/42.231.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER
Content-Type: application/sdp
Content-Length: 384

v=0
o=3001 5004 138 IN IP4 192.168.6.147
s=Mapping
c=IN IP4 192.168.6.147
t=0 0
m=audio 5004 RTP/AVP 8 0 9 96 97 2 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (15 headers 17 lines) ---
Sending to 192.168.6.147:5060 (no NAT)
Using INVITE request as basis request - 2522539599@192_168_6_147
Found peer '3001' for '3001' from 192.168.6.147:5060

останавливается на 10 сек, и потом продолжает уже в нормальном режиме:

Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G722 for ID 9
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|g726|alaw|g722|g729|g726aal2)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.6.147:5004
Looking for 3002 in common (domain 192.168.1.129)
sip_route_dump: route/path hop: <sip:3001@192.168.6.147:5060>

<--- Transmitting (no NAT) to 192.168.6.147:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.6.147:5060;branch=z9hG4bKf3ae399d6c8efe4097e73c7d70d1c104;received=192.168.6.147;rport=5060
From: "3001" <sip:3001@192.168.1.129>;tag=2038128166
To: <sip:3002@192.168.1.129;user=phone>
Call-ID: 2522539599@192_168_6_147
CSeq: 3 INVITE
Server: MSK-AST-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:3002@192.168.1.129:5060>
Content-Length: 0
...
...
...

То есть остановка идете перед

Found RTP audio format 8

По каким причинам он может здесь тормозить? Заранее спасибо