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我试图在使用PJSIP的iOS应用程序中获取SRTP。我有TLS工作，没有SRTP我可以拨打和接听电话。但是，对于SRTP，我在INVITE上收到了这个奇怪的488错误。它无法初始化媒体。SRTP问题：PJSIP初始化媒体通道时出错：此处不可接受[status = 170488]
我读过关于编解码器的其他文章。但是我已经确保Asterisk服务器使用的代码和我的iOS应用程序使用PJSIP库编译的代码是相同的。我在这里看到的唯一的事情是我启用了加密并且PJSIP不喜欢它。有什么想法吗？
INVITE sip:[REDACTED]@[REDACTED]:47229;transport=TLS;ob SIP/2.0
Via: SIP/2.0/TLS [REDACTED]:5161;rport;branch=z9hG4bKPj8ea1a332-0748-438f-ae74-5d17b038891d;alias
From: "Test" <sip:[email protected]>;tag=7c3663cb-b5f5-4762-8526-8425d18b2466
To: <sip:[REDACTED]@[REDACTED];ob>
Contact: <sip:[email protected][REDACTED]:5161;transport=TLS>
Call-ID: f454ef36-01ea-4f29-9482-4a10768bf1b7
CSeq: 24942 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, path
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-AsteriskNOW-13.0.190.12(13.13.1)
Content-Type: application/sdp
Content-Length: 398
v=0
o=- 1582453973 1582453973 IN IP4 172.31.18.138
s=Asterisk
c=IN IP4 [REDACTED]
t=0 0
m=audio 11410 RTP/AVP 3 110 9 97 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:84m7hqGvMjTU21xzkhBS3RQpQQjJ+aep0VwSlhx+
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
--end msg--
19:10:11.601 pjsua_call.c .Incoming Request msg INVITE/cseq=24942 (rdata0x1421f0540)
19:10:11.603 tsx0x1421fe0a8 ...Transaction created for Request msg INVITE/cseq=24942 (rdata0x1421f0540)
19:10:11.603 tsx0x1421fe0a8 ..Incoming Request msg INVITE/cseq=24942 (rdata0x1421f0540) in state Null
19:10:11.603 tsx0x1421fe0a8 ...State changed from Null to Trying, event=RX_MSG
19:10:11.603 dlg0x1421fd8a8 ....Transaction tsx0x1421fe0a8 state changed to Trying
19:10:11.603 dlg0x1421fd8a8 ..UAS dialog created
19:10:11.603 dlg0x1421fd8a8 ..Module mod-invite added as dialog usage, data=0x141de7588
19:10:11.603 dlg0x1421fd8a8 ...Session count inc to 3 by mod-invite
19:10:11.603 inv0x1421fd8a8 ..UAS invite session created for dialog dlg0x1421fd8a8
19:10:11.603 dlg0x1421fd8a8 ...Session count inc to 3 by mod-pjsua
19:10:11.603 pjsua_media.c ..Call 0: initializing media..
19:10:11.603 pjsua_call.c ..Error initializing media channel: Not Acceptable Here [status=170488]
19:10:11.604 endpoint ..Response msg 488/INVITE/cseq=24942 (tdta0x1421fe800) created
19:10:11.604 dlg0x1421fd8a8 ...Sending Response msg 488/INVITE/cseq=24942 (tdta0x1421fe800)
19:10:11.606 tsx0x1421fe0a8 ...Sending Response msg 488/INVITE/cseq=24942 (tdta0x1421fe800) in state Trying
19:10:11.606 pjsua_core.c ....TX 429 bytes Response msg 488/INVITE/cseq=24942 (tdta0x1421fe800) to TLS [REDACTED]:5161:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS [REDACTED]:5161;rport=5161;received=[REDACTED];branch=z9hG4bKPj8ea1a332-0748-438f-ae74-5d17b038891d;alias
Call-ID: f454ef36-01ea-4f29-9482-4a10768bf1b7
From: "Test" <sip:[email protected]>;tag=7c3663cb-b5f5-4762-8526-8425d18b2466
To: <sip:[REDACTED]@[REDACTED];ob>;tag=5oFGceZO4ZaKpLFEg7piOrM7IV2yeDLT
CSeq: 24942 INVITE
Content-Length: 0
--end msg--