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linphone iphone sdk를 사용하여 IOS VOIP 응용 프로그램을 개발 중입니다.누락 된 오디오/비디오 SDP 문제
또한 VOIP 서버에 Asterisk PBX 12.6.1을 사용했습니다.
나는 B.
내가 별표 서버에서 SIP 로그를 확인하고 정확하게 별표 오디오 만 SDP를 가지고A에서 음성 통화를했다.
하지만 NAT를 통해 피어에 초대 메시지를 보내면 Invite 메시지의 비디오 SDP가 표시됩니다.
이것은 초대 메시지가 NAT를 통과하면 비디오 SDP가 초대 메시지에 추가되었음을 의미합니다.
그래서 음성 전화를 걸었지만 현재 화상 SDP가 있기 때문에 화상 회의라는 것을 알았습니다.
어떻게 솔루션을 찾을 수 있습니까?
감사합니다.
다음은 SIP 로그입니다.
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:63787;branch=z9hG4bK.g2B2Eb9kT;rport
From: "8615524541202" <sip:[email protected]>;tag=Z9ZmQR1UW
To: "xiu20170508" <sip:[email protected]>
CSeq: 20 INVITE
Call-ID: TmNmS3XX2H
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 763
Contact: <sip:[email protected]:63787;transport=udp>;+sip.instance="<urn:uuid:ccc94acb-59d0-4048-8ccf-4795a8510248>"
User-Agent: iPhone.6.Plus_iOS10.3.3/3.16.4-83-g47dba0d2 (belle-sip/1.6.1)
v=0
o=8615524541202 4023 1235 IN IP4 NATIP
s=Talk
c=IN IP4 NATIP
b=AS:512
t=0 0
a=ice-pwd:2aa3fb71d33aaeabd9b709c5
a=ice-ufrag:11ed9d27
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7268 RTP/AVPF 96 0 8 18 101 97
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:97 telephone-event/8000
a=candidate:1 1 UDP 2130706431 192.168.1.101 7268 typ host
a=candidate:1 2 UDP 2130706430 192.168.1.101 7269 typ host
a=candidate:2 1 UDP 1694498815 NATIP 7268 typ srflx raddr 192.168.1.101 rport 7268
a=candidate:2 2 UDP 1694498814 NATIP 7269 typ srflx raddr 192.168.1.101 rport 7269
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
Found RTP audio format 96
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 97
Found audio description format opus for ID 96
Found unknown media description format telephone-event for ID 101
Found audio description format telephone-event for ID 97
Capabilities: us - (slin|g729|speex|h264|opus|vp8), peer - audio=(ulaw|alaw|g729|opus)/video=(nothing)/text=(nothing), combined - (g729|opus)
Reliably Transmitting (NAT) to NATIP:38594:
INVITE sip:[email protected]:38594;transport=udp SIP/2.0
Via: SIP/2.0/UDP myvoip.com:5060;branch=z9hG4bK291104a5;rport
Max-Forwards: 70
From: "8615524541202" <sip:[email protected]>;tag=as112aacc0
To: <sip:[email protected]:38594;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.6.1
Date: Tue, 05 Dec 2017 14:26:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1454
v=0
o=root 533584201 533584201 IN IP4 myvoip.com
s=Asterisk PBX 12.6.1
c=IN IP4 myvoip.com
b=CT:384
t=0 0
m=audio 10156 RTP/AVP 96 110 10 18 97
a=rtpmap:96 opus/48000/2
a=fmtp:96 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:110 speex/8000
a=rtpmap:10 L16/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:60
a=ice-ufrag:1849182226f511bd0c82b340773eb664
a=ice-pwd:4078b4a75c3730c13ed92aa85e317bce
a=candidate:Hac1f1be6 1 UDP 2130706431 stunip 10156 typ host
a=candidate:S3642e9ba 1 UDP 1694498815 myvoip.com 10156 typ srflx
a=candidate:Hac1f1be6 2 UDP 2130706430 stunip 10157 typ host
a=candidate:S3642e9ba 2 UDP 1694498814 myvoip.com 10157 typ srflx
a=sendrecv
m=video 13866 RTP/AVP 100 99
a=ice-ufrag:364315c202baf54c125f661f1f1a1bf1
a=ice-pwd:50974e571f2e161023905c2540f88ec0
a=candidate:Hac1f1be6 1 UDP 2130706431 stunip 13866 typ host
a=candidate:S3642e9ba 1 UDP 1694498815 myvoip.com 13866 typ srflx
a=candidate:Hac1f1be6 2 UDP 2130706430 stunip 13867 typ host
a=candidate:S3642e9ba 2 UDP 1694498814 myvoip.com 13867 typ srflx
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv