通过FreeSWITCH，Linphone呼叫WebRTC的视频不通问题解决

最新推荐文章于 2024-08-27 16:39:27 发布

转载最新推荐文章于 2024-08-27 16:39:27 发布 · 2.4k 阅读

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原文链接：https://my.oschina.net/andywang1988/blog/1186669

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本文探讨了FreeSWITCH在处理不同类型的WebRTC媒体流时的行为差异，特别是针对TCP与WebSocket注册方式的不同处理逻辑。文中还讨论了如何在非WebSocket连接中启用ICE协商及DTLS加密的问题。

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账号A（Linphone的IOS客户端）呼叫账号B，相同的呼叫操作操作。

如果账号B通过linphone的TCP注册到FS，FS作为B-leg会发送过来的INVITE消息中的SDP为：

2017-07-05 10:18:53.313008 [DEBUG] mod_sofia.c:90 sofia/internal/3001@172.18.8.14:5060 SOFIA INIT
2017-07-05 10:18:53.313008 [DEBUG] sofia_glue.c:1253 sip:3001@223.93.174.216:48983;transport=tcp Setting proxy route to sofia/internal/3001@172.18.8.14:5060
2017-07-05 10:18:53.313008 [DEBUG] sofia_glue.c:1282 sofia/internal/3001@172.18.8.14:5060 sending invite version: 1.6.12  64bit
Local SDP:
v=0
o=FreeSWITCH 1499189835 1499189836 IN IP4 120.27.248.33
s=FreeSWITCH
c=IN IP4 120.27.248.33
t=0 0
m=audio 31298 RTP/AVP 102 0 8 103 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 telephone-event/48000
a=fmtp:103 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 28250 RTP/AVP 96
b=AS:256
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 ccm tmmbr
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli

账户B使用sipML的WebSocket注册到FS时，收到的FS发送过来的INVITE消息中SDP为：

2017-07-05 10:18:00.433013 [DEBUG] sofia_glue.c:1282 sofia/internal/3006@df7jal23ls0d.invalid 	 invite version: 1.6.12  64bit
Local SDP:
v=0
o=FreeSWITCH 1499193706 1499193707 IN IP4 120.27.248.33
s=FreeSWITCH
c=IN IP4 120.27.248.33
t=0 0
a=msid-semantic: WMS um09FCvgZw0FRK8COZAZEknMMgVkJrC3
m=audio 27374 RTP/SAVPF 102 0 8 103 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 telephone-event/48000
a=rtpmap:101 telephone-event/8000
a=fingerprint:sha-256 F1:00:B2:56:60:59:61:BF:27:B3:1C:CE:AF:E6:AB:79:CF:E1:04:36:CC:01:28:47:D5:F4:64:E6:6C:42:96:34
a=setup:actpass
a=rtcp-mux
a=rtcp:27374 IN IP4 120.27.248.33
a=ssrc:291447064 cname:Es39VogLL7gtHfVm
a=ssrc:291447064 msid:um09FCvgZw0FRK8COZAZEknMMgVkJrC3 a0
a=ssrc:291447064 mslabel:um09FCvgZw0FRK8COZAZEknMMgVkJrC3
a=ssrc:291447064 label:um09FCvgZw0FRK8COZAZEknMMgVkJrC3a0
a=ice-ufrag:4mczwqc9wXj5fLZJ
a=ice-pwd:4SIP7PqIMx2jCOwx0yXmOZla
a=candidate:5447911826 1 udp 659136 120.27.248.33 27374 typ host generation 0
a=candidate:5447911826 2 udp 659136 120.27.248.33 27374 typ host generation 0
a=ptime:20
a=sendrecv
m=video 16960 RTP/SAVPF 96
b=AS:256
a=rtpmap:96 VP8/90000
a=fingerprint:sha-256 F1:00:B2:56:60:59:61:BF:27:B3:1C:CE:AF:E6:AB:79:CF:E1:04:36:CC:01:28:47:D5:F4:64:E6:6C:42:96:34
a=setup:actpass
a=rtcp-mux
a=rtcp:16960 IN IP4 120.27.248.33
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 ccm tmmbr
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=ssrc:3836824708 cname:Es39VogLL7gtHfVm
a=ssrc:3836824708 msid:um09FCvgZw0FRK8COZAZEknMMgVkJrC3 v0
a=ssrc:3836824708 mslabel:um09FCvgZw0FRK8COZAZEknMMgVkJrC3
a=ssrc:3836824708 label:um09FCvgZw0FRK8COZAZEknMMgVkJrC3v0
a=ice-ufrag:PYA2DLzBmNZZPqwR
a=ice-pwd:mQ5PCkDyTZVUmZq2NQVhskJE
a=candidate:5820208970 1 udp 659136 120.27.248.33 16960 typ host generation 0
a=candidate:5820208970 2 udp 659135 120.27.248.33 16960 typ host generation 0
a=end-of-candidates

非WS注册到FreeSWITCH上，呼叫时，需要指定"media_webrtc=true",才会产生ICE相关的信息

    <action application="export" data="media_webrtc=true"/>                                                                                                                                                                                 
    <!--action application="export" data="add_ice_candidates=true"/-->
    <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>

FreeSwitch源码中src/switch_rtp.c中，默认webrtc相关的媒体流时加密的DTLS方式的，如果你建立的UDP不是加密的，FS不会进行转发到另一端。这个逻辑判断是根据WebRTC的W3C标准来的。我们自己不需要加密，所以，我把这个逻辑判断给屏蔽了。

static int rtp_write_ready(switch_rtp_t *rtp_session, uint32_t bytes, int line)
{
    if (!rtp_session) return 0;
 
    if (rtp_session->ice.ice_user && !(rtp_session->ice.rready || rtp_session->ice.ready)) {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG3, "Skip sending %s packet %ld bytes (ice not ready @ line %d!)\n",
                          rtp_type(rtp_session), (long)bytes, line);
        return 0;
    }
 
#if 0//wangweichao add for support pjsip with webrtc --20170706---                                                                                                                                                                          
 
    if (rtp_session->dtls && rtp_session->dtls->state != DS_READY) {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG3, "Skip sending %s packet %ld bytes (dtls not ready @ line %d!)\n",
                          rtp_type(rtp_session), (long)bytes, line);
 
        return 0;
    }
#endif
    return 1;
}

转载于:https://my.oschina.net/andywang1988/blog/1186669