我们有一个门电话，通过RTSP从IP摄像机获取视频，然后进行视频通话转发视频 在很多情况下我工作得很好，但是，有些时候，PBX用Port = 0 sdp拒绝它
来自我们产品的INVITE（带有sdp）和来自PBX的200 OK
重要的事情我忘记在发布时将其纳入问题
当我从带显示器的电话呼叫到门电话时，它工作正常 但没有从门电话到“内部”
也许这与门电话呼叫铃声组的事实有关？ （PBX是Grandstream UCM6102）
如果有人可以提供帮助，我会非常感激
INVITE
INVITE sip:2234@10.1.10.79 SIP/2.0
Via: SIP/2.0/UDP 10.1.200.210:5060;branch=z9hG4bk-3536322826-491;rport
Max-Forwards: 70
Contact: <sip:2285@10.1.200.210:5060>
To: <sip:2234@10.1.10.79>
From: <sip:2285@10.1.10.79>;tag=4128451792
Call-ID: 2863496175@10.1.200.210
CSeq: 10 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY
Supported: replaces
Content-Type: application/sdp
User-Agent: OEM WANPAGE V3.11.2 w9
Content-Length: 306
v=0
o=- 4546474849 5556575859 IN IP4 10.1.200.210
s=OEM WANPAGE V3.11.2
c=IN IP4 10.1.200.210
t=0 0
m=audio 55000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 50000 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
来自PBX的200 OK
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.200.210:5060;branch=z9hG4bk-3536322826-984;received=10.1.200.210;rport=5060
From: <sip:2285@10.1.10.79>;tag=4128451792
To: <sip:2234@10.1.10.79>;tag=as38b54d5f
Call-ID: 2863496175@10.1.200.210
CSeq: 11 INVITE
Server: GHJ-2.11.0(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:2234@82.81.161.139:5060>
P-Asserted-Identity: "intercom" <sip:2234@10.1.10.79>
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 648544421 648544421 IN IP4 82.81.161.139
s=Asterisk PBX 11.21.2
c=IN IP4 82.81.161.139
t=0 0
m=audio 15676 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99
另一个例子
INVITE
INVITE sip:2234@10.1.10.79 SIP/2.0
Via: SIP/2.0/UDP 10.1.200.210:5060;branch=z9hG4bk-3536322826-3;rport
Max-Forwards: 70
Contact: <sip:2285@10.1.200.210:5060>
To: <sip:2234@10.1.10.79>
From: <sip:2285@10.1.10.79>;tag=3603104825
Call-ID: 2338149257@10.1.200.210
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY
Supported: replaces
Content-Type: application/sdp
User-Agent: OEM WANPAGE V3.11.2 w9
Content-Length: 372
v=0
o=- 4546474849 5556575859 IN IP4 10.1.200.210
s=OEM WANPAGE V3.11.2
c=IN IP4 10.1.200.210
t=0 0
m=audio 55000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 50000 RTP/AVP 99
b=AS:256
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=420014
a=sendrecv
200 OK
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.200.210:5060;branch=z9hG4bk-3536322826-8;received=10.1.200.210;rport=5060
From: <sip:2285@10.1.10.79>;tag=3603104825
To: <sip:2234@10.1.10.79>;tag=as606d193f
Call-ID: 2338149257@10.1.200.210
CSeq: 4 INVITE
Server: GHJ-2.11.0(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:2234@82.81.161.139:5060>
P-Asserted-Identity: "intercom" <sip:2234@10.1.10.79>
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 1380160206 1380160206 IN IP4 82.81.161.139
s=Asterisk PBX 11.21.2
c=IN IP4 82.81.161.139
t=0 0
m=audio 15342 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99