星号WebRTC传出呼叫延迟

Question

我运行一个星号16安装和一个基于WebPhone的SIP.js。不幸的是，当我给别人打电话的时候，我常常听不到头几秒钟的声音。但是来电一切都很好。

星号位于数据中心，browser / client位于NAT之后。

日志(请参阅秒11至13之间的延迟)。

[Nov 2 17:58:11] VERBOSE[15217][C-00000002] app_dial.c: PJSIP/hativ-voip-00000003 answered PJSIP/hativ-00000002
[Nov 2 17:58:11] VERBOSE[15226][C-00000002] bridge_channel.c: Channel PJSIP/hativ-voip-00000003 joined 'simple_bridge' basic-bridge <80f71862-7910-4363-97e4-8d8a9e98765f>
[Nov 2 17:58:11] VERBOSE[15217][C-00000002] bridge_channel.c: Channel PJSIP/hativ-00000002 joined 'simple_bridge' basic-bridge <80f71862-7910-4363-97e4-8d8a9e98765f>
[Nov 2 17:58:13] VERBOSE[15217][C-00000002] res_rtp_asterisk.c: 0x7f05cc0773a0 -- Strict RTP qualifying stream type: audio
[Nov 2 17:58:13] VERBOSE[15217][C-00000002] res_rtp_asterisk.c: 0x7f05cc0773a0 -- Strict RTP switching source address to 91.67.195.16:58920
[Nov 2 17:58:13] VERBOSE[15217][C-00000002] res_rtp_asterisk.c: 0x7f05cc0773a0 -- Strict RTP learning complete - Locking on source address 91.67.195.16:58920
[Nov 2 17:58:13] VERBOSE[15226][C-00000002] res_rtp_asterisk.c: 0x7f05cc0860d0 -- Strict RTP switching to RTP target address 212.117.203.158:32406 as source
[Nov 2 17:58:13] VERBOSE[15226][C-00000002] res_rtp_asterisk.c: 0x7f05cc0860d0 -- Strict RTP learning complete - Locking on source address 212.117.203.158:32406

星号帐户的端点配置(用于连接到星号)：

type=endpoint

dtls_ca_file=/etc/asterisk/certificates/chain.pem
dtls_cert_file=/etc/asterisk/certificates/cert.pem
dtls_private_key=/etc/asterisk/certificates/privkey.pem

direct_media=no
force_rport=yes
rtp_ipv6=yes
rtp_symmetric=yes

; Sets dtls_setup=actpass, dtls_verify=fingerprint, ice_support=yes, media_encryption=dtls, media_use_received_transport=yes, rtcp_mux=yes, use_avpf=yes
webrtc=yes

disallow=all
allow=opus
allow=alaw
allow=ulaw

SIP帐户的端点配置(用于连接到SIP提供程序)：

type=endpoint

context=incoming
dtmf_mode=rfc4733
force_rport=yes
language=de
rewrite_contact=yes
rtp_symmetric=yes
timers=yes

from_domain=example.com

disallow=all
allow=alaw

有人知道是什么原因造成了延误吗？

Answer 1

尝试删除sip.conf上星号端点上的进步带183：

progressinband=never

这将不会发送183和避免RTP环没有证书交换。