如何从sipML5中的SIP邀请中删除不必要的数据？ 现在它发送到我的服务器时太大了（只需要音频）。它将接受最多1,500个字节，并且必须在UDP上。
你能告诉我怎么做吗？如何删除一些编解码器等？我对此一无所知，只是现在学习sipML。在其他帖子中有像删除编解码器这样的答案，但没有答案如何做到这一点：）
我的邀请：
SEND: INVITE sip:some_client@some_address SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKRyhpzJOIUVBDwgMLxDIq1CAmXFZo2HkD;rport
From: <sip:some_client@some_address>;tag=hpGTFTQ0Kpt6JFgsn8Bc
To: <sip:some_number_to_call@some_address>
Contact: "undefined"<sip:some_client@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
Call-ID: 558d80b1-b383-344e-dea1-b95ddbe9dc3f
CSeq: 30366 INVITE
Content-Type: application/sdp
Content-Length: 2247
Max-Forwards: 70
v=0
o=- 3717615351353762000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS HMFRvujnzsUIWdP6940nngmFOxrtVbMeG8nr
m=audio 52548 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 91.241.5.1
a=rtcp:11803 IN IP4 91.241.5.1
a=candidate:2002913928 1 udp 2122260223 192.168.81.65 52691 typ host generation 0
a=candidate:2002913928 2 udp 2122260222 192.168.81.65 58663 typ host generation 0
a=candidate:4129950780 1 udp 1686052607 91.241.5.1 52548 typ srflx raddr 192.168.81.65 rport 52691 generation 0
a=candidate:4129950780 2 udp 1686052606 91.241.5.1 11803 typ srflx raddr 192.168.81.65 rport 58663 generation 0
a=candidate:971110008 1 tcp 1518280447 192.168.81.65 9 typ host tcptype active generation 0
a=candidate:971110008 2 tcp 1518280446 192.168.81.65 9 typ host tcptype active generation 0
a=candidate:4129950780 1 udp 1686052607 91.241.5.1 44693 typ srflx raddr 192.168.81.65 rport 52691 generation 0
a=candidate:4129950780 2 udp 1686052606 91.241.5.1 47874 typ srflx raddr 192.168.81.65 rport 58663 generation 0
a=candidate:4129950780 1 udp 1686052607 91.241.5.1 22880 typ srflx raddr 192.168.81.65 rport 52691 generation 0
a=candidate:4129950780 2 udp 1686052606 91.241.5.1 19665 typ srflx raddr 192.168.81.65 rport 58663 generation 0
a=ice-ufrag:FOoiTf25RFgO/bOx
a=ice-pwd:ANc8oBQwW5zwBMd9lK2slJNN
a=fingerprint:sha-256 78:A2:A4:13:11:A2:74:25:6E:B8:D4:9E:F3:1B:71:7E:A5:10:38:39:01:CC:93:C1:74:B3:96:25:71:C8:D2:5D
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1519941870 cname:s8rO7shwZbHh0wyw
a=ssrc:1519941870 msid:HMFRvujnzsUIWdP6940nngmFOxrtVbMeG8nr 38192f79-8fcc-45c9-b812-55e01c26364e
a=ssrc:1519941870 mslabel:HMFRvujnzsUIWdP6940nngmFOxrtVbMeG8nr
a=ssrc:1519941870 label:38192f79-8fcc-45c9-b812-55e01c26364e
答案 0 :(得分：0)
建议不要在JavaScript的客户端修改它。您的服务器或WebRTC到SIP网关应自动解决此问题（通过TCP转发或删除不必要的行，如果只能在UDP上转发）。