Собсно пытаюсь тут поэксперементировать с сабжем, телефоны Yealink T22P. SIP на TLS настроил без проблем, телефоны регистрируются по TLS, звонки ходят. В телефонах включаю SRTP, на экстеншенах ставлю encryption=yes, но в итоге имею «488 Not acceptable here» при попытке позвонить:
INVITE sip:1797@sip.domain.ru:5061 SIP/2.0
Via: SIP/2.0/TLS 10.1.3.169:17315;branch=z9hG4bK1937264885
From: "Test" <sip:1951@sip.domain.ru:5061>;tag=1027830000
To: <sip:1797@sip.domain.ru:5061>
Call-ID: 1422776310@10.1.3.169
CSeq: 2 INVITE
Contact: <sip:1951@10.1.3.169:17315;transport=TLS>
Authorization: Digest username="1951", realm="asterisk", nonce="65eee7be", uri="sip:1797@sip.domain.ru:5061", response="d1ffcc4667693ac7f72d690d7270aaa9", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T22P 7.71.14.5
Supported: replaces,timer
Min-SE: 90
Session-Expires: 90
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 443
v=0
o=- 20003 20003 IN IP4 10.1.3.169
s=SDP data
c=IN IP4 10.1.3.169
t=0 0
m=audio 11786 RTP/AVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTNlYzM2ZGIzYmE0YTBlMGY4MmIyMTAANzA2OTA0
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzdmYzEwYWM0N2M4ZTJkADNjZDcyYzkANDhhZTUw
a=crypto:3 F8_128_HMAC_SHA1_80 inline:MzEyMjViMTdiMzcwOWI1ADQ2NjI2MDA0ZWM0ODIy
a=rtpmap:0 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<--- Reliably Transmitting (NAT) to 10.1.3.169:17315 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS 10.1.3.169:17315;branch=z9hG4bK1937264885;received=10.1.3.169;rport=17315
From: "Test" <sip:1951@sip.domain.ru:5061>;tag=1027830000
To: <sip:1797@sip.domain.ru:5061>;tag=as3ef74fd0
Call-ID: 1422776310@10.1.3.169
CSeq: 2 INVITE
Server: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
ЧЯДНТ?
