问题描述
我可以从我的网络客户端注册到我的 freeswitch.但是，当我尝试拨打电话时，电话被拒绝，488 在这里不可接受.来自我正在获取的 freeswitch 控制台日志.
I can register from my webclient to my freeswitch. But, when I try to make call the call gets rejected with 488 not acceptable here. From freeswitch console log im getting.
2014-07-22 22:03:59.673585 [DEBUG] switch_core_state_machine.c:53 sofia/internal/[email protected] Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
我加了<action application="export" data="rtp_secure_media=true"/>用我的分机；但没有运气.
I added < action application="export" data="rtp_secure_media=true" />with my extension; but no luck.
下面是我的INVITE的SDP
below is the SDP of my INVITE
v=0
o=Mozilla-SIPUA-31.0 26508 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:13497e25
a=ice-pwd:515d61f08d909117e022674f3dce748e
a=fingerprint:sha-256 2E:CF:7E:8F:EC:1A:F4:B1:D3:CF:39:C3:8A:A0:D0:53:B3:46:00:D0:93:46:53:29:AB:B7:03:83:39:FB:23:32
m=audio 55760 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 184.69.59.132
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.16.1.188 55760 typ host
a=candidate:1 1 UDP 1692467199 184.69.59.132 55760 typ srflx raddr 172.16.1.188 rport 55760
a=candidate:5 1 UDP 2128543999 192.168.56.1 55761 typ host
a=candidate:10 1 UDP 2128478463 192.168.232.1 55762 typ host
a=candidate:15 1 UDP 2128412927 192.168.146.1 55763 typ host
a=candidate:0 2 UDP 2128609534 172.16.1.188 55764 typ host
a=candidate:1 2 UDP 1692467198 184.69.59.132 55764 typ srflx raddr 172.16.1.188 rport 55764
a=candidate:5 2 UDP 2128543998 192.168.56.1 55765 typ host
a=candidate:10 2 UDP 2128478462 192.168.232.1 55766 typ host
a=candidate:15 2 UDP 2128412926 192.168.146.1 55767 typ host
a=rtcp-mux
以下是我从 freeswitch 中获得的编解码器列表.我没有安装 opus，但我有 G711 ulaw 和 alaw
Below is my codec lists from freeswitch. I dont have opus installed, but I do have G711 ulaw and alaw
show codecs
type,name,ikey
codec,ADPCM (IMA),mod_voipcodecs
codec,AMR,mod_amr
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_voipcodecs
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_voipcodecs
codec,G.726 16k (AAL2),mod_voipcodecs
codec,G.726 24k,mod_voipcodecs
codec,G.726 24k (AAL2),mod_voipcodecs
codec,G.726 32k,mod_voipcodecs
codec,G.726 32k (AAL2),mod_voipcodecs
codec,G.726 40k,mod_voipcodecs
codec,G.726 40k (AAL2),mod_voipcodecs
codec,G.729,mod_g729
codec,GSM,mod_voipcodecs
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_voipcodecs
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,Polycom(R) G722.1/G722.1C,mod_siren
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
codec,iLBC,mod_ilbc
可能是什么问题?
推荐答案
根据日志，这也可能是与 m 行的兼容性问题，其中 m 行将所有传输协议列在了一起.
Based on the logs it could also be a compatibilty issue with the m line that has all the transport protocols listed together.
UDP/TLS/RTP/SAVPF.这可能会受到这些线程中提到的兼容性问题的影响.也许您可以尝试将其限制为更简单的形式，并在可能的情况下尝试.
UDP/TLS/RTP/SAVPF. This can be subject to compatibility issues as mentioned in these threads. Maybe you could try to restrict it to the simpler form and try it if possible.
https://code.google.com/p/webrtc/问题/详细信息?id=2796
http://lists.freeswitch.org/pipermail/freeswitch-users/2013-July/097617.html
这篇关于Freeswitch 和 webRTC:媒体被 488 拒绝的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持！