Asterisk无法在LTE(4G)网络上传送声音

时间:2014-04-16 05:42:48

标签: sip asterisk 4g

我已经安装了Asterisk 11,两个wifi手机可以通过星号服务器进行通话。然而,wifi电话和LTE(4G)电话无法发出声音。

Asterisk sip.conf

[general]
context=default                       ; Default context for incoming calls
bindport=5060                   ; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0           ; IP address to bind to (0.0.0.0 binds to all)
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=alaw
register => 12121111111:1234:11111111@sipauth.deltathree.com/1000
srvlookup=no
directrtpsetup=yes
trustpid=yes
sendrpid=no
qualify=yes
callevents=yes
insecure=invite
pedantic=no
videosupport=yes
canreinvite=yes
nat=yes
externip=XXX.XXX.91.12
localnet=10.7.21.4/255.255.255.0
qualify=yes
directmedia=yes

啜饮设置

Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       Yes
  User Agent:             Asterisk PBX 11.8.1
  SDP Session Name:       Asterisk PBX 11.8.1
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            On
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Enabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Enabled using externhost
  Externhost:             XXX.52.91.12:0
  Externaddr:             XXX.52.91.12:0
  Externrefresh:          600
  Localnet:               XX.7.21.0/255.255.255.0
                          XX.7.21.0/255.255.255.0

Global Signalling Settings:
---------------------------
  Codecs:                 (ulaw|alaw)
  Codec Order:            ulaw:20,alaw:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Sub. min duration       60 secs
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                default
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                2000
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk

Realtime SIP Settings:
----------------------
  Realtime Peers:         Yes
  Realtime Regs:          No
  Cache Friends:          No
  Update:                 Yes
  Ignore Reg. Expire:     No
  Save sys. name:         No
  Auto Clear:             120 (Disabled)

啜饮日志 当我看着sip日志时,它看起来很好。我只看到一个&#34;邀请&#34;从服务器到wifi电话。

interface: eth0 (10.7.21.0/255.255.255.0)
filter: ( port 5060 ) and (ip or ip6)
#
U 2014/04/16 22:46:28.514023 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
INVITE sip:2000@asterisk-sip-domain.com SIP/2.0.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;rport.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>.
CSeq: 20 INVITE.
Call-ID: Z6lXHBKOyd.
Max-Forwards: 70.
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 372.
Contact: <sip:1000@//WIFI-PUBLIC-IP//:1495>;+sip.instance="<urn:uuid:41bf1699-9e9a-4817-8b8c-e51f7b4ae2dc>".
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
.
v=0.
o=1000 2350 2859 IN IP4 //WIFI-PRIVATE-IP//.
s=Talk.
c=IN IP4 //WIFI-PRIVATE-IP//.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 124 120 111 110 0 8 101.
a=rtpmap:124 opus/48000.
a=fmtp:124 useinbandfec=1; usedtx=1.
a=rtpmap:120 SILK/16000.
a=rtpmap:111 speex/16000.
a=fmtp:111 vbr=on.
a=rtpmap:110 speex/8000.
a=fmtp:110 vbr=on.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2014/04/16 22:46:28.517399 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>.
Call-ID: Z6lXHBKOyd.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Content-Length: 0.
.

#
U 2014/04/16 22:46:28.522887 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
INVITE sip:2000@//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
Max-Forwards: 70.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>.
Contact: <sip:1000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:28 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 258.
.
v=0.
o=root 1526682879 1526682879 IN IP4 //WIFI-PRIVATE-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //WIFI-PRIVATE-IP//.
t=0 0.
m=audio 7076 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.

#
U 2014/04/16 22:46:29.022450 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
INVITE sip:2000@//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
Max-Forwards: 70.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>.
Contact: <sip:1000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:28 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 258.
.
v=0.
o=root 1526682879 1526682879 IN IP4 //WIFI-PRIVATE-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //WIFI-PRIVATE-IP//.
t=0 0.
m=audio 7076 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.

#
U 2014/04/16 22:46:29.113047 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: sip:2000@//LTE-PHONE-PUBLIC-IP//:63968.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
.

#
U 2014/04/16 22:46:29.426139 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.

#
U 2014/04/16 22:46:29.426158 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.

#
U 2014/04/16 22:46:29.427976 f:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
Call-ID: Z6lXHBKOyd.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Content-Length: 0.
.

** (WHY IT MAKES ONE MORE INVITE FROM SERVER TO WIFI-PHONE???)**
# 
U 2014/04/16 22:46:30.307448 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:48504
INVITE sip:1000@//WIFI-PUBLIC-IP//:48504 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK451726cf;rport.
Max-Forwards: 70.
From: <sip:2000@//AMAZON-EC2-SERVER//>;tag=as30b8a8a5.
To: <sip:1000@//WIFI-PUBLIC-IP//:48504>.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 1c2fd2cd6a4ac372408845e8077ba2b5@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:14 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 741350827 741350827 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
t=0 0.
m=audio 30390 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.

#
U 2014/04/16 22:46:30.816230 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Contact: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;+sip.instance="<urn:uuid:8afceca3-368f-4f57-a586-6056d3492371>".
Content-Type: application/sdp.
Content-Length: 183.
.
v=0.
o=2000 2310 1562 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Talk.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
b=AS:380.
t=0 0.
m=audio 30390 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2014/04/16 22:46:30.816888 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
ACK sip:2000@//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK680dd0d2;rport.
Max-Forwards: 70.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Contact: <sip:1000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/16 22:46:30.817278 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 200 OK.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
Call-ID: Z6lXHBKOyd.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=root 1551912347 1551912347 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
t=0 0.
m=audio 30390 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.

#
U 2014/04/16 22:46:30.925455 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
ACK sip:2000@//AMAZON-EC2-SERVER//:5060 SIP/2.0.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;rport;branch=z9hG4bK.qV8rz6rI4.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
CSeq: 20 ACK.
Call-ID: Z6lXHBKOyd.
Max-Forwards: 70.
.

#
U 2014/04/16 22:46:35.277987 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
BYE sip:1000@//AMAZON-EC2-SERVER//:5060 SIP/2.0.
Via: SIP/2.0/UDP //LTE-PHONE-PUBLIC-IP//:63968;branch=z9hG4bK.Jfn1vpiLT;rport.
From: <sip:2000@//LTE-PHONE-PUBLIC-IP//>;tag=zZBSo25.
To: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
CSeq: 111 BYE.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
Max-Forwards: 70.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
.

#
U 2014/04/16 22:46:35.278525 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
SIP/2.0 200 OK.
Via: SIP/2.0/UDP //LTE-PHONE-PUBLIC-IP//:63968;branch=z9hG4bK.Jfn1vpiLT;received=//LTE-PHONE-PUBLIC-IP//;rport=63968.
From: <sip:2000@//LTE-PHONE-PUBLIC-IP//>;tag=zZBSo25.
To: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 111 BYE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.

#
U 2014/04/16 22:46:35.278797 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
INVITE sip:1000@//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
Max-Forwards: 70.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 1551912347 1551912348 IN IP4 //AMAZON-EC2-SERVER//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //AMAZON-EC2-SERVER//.
t=0 0.
m=audio 19500 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.

#
U 2014/04/16 22:46:35.418765 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
.

#
U 2014/04/16 22:46:35.441248 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Contact: <sip:1000@//WIFI-PUBLIC-IP//:1495>;+sip.instance="<urn:uuid:41bf1699-9e9a-4817-8b8c-e51f7b4ae2dc>".
Content-Type: application/sdp.
Content-Length: 180.
.
v=0.
o=1000 2350 2861 IN IP4 //WIFI-PRIVATE-IP//.
s=Talk.
c=IN IP4 //WIFI-PRIVATE-IP//.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2014/04/16 22:46:35.441661 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
ACK sip:1000@//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK7dd12d7d;rport.
Max-Forwards: 70.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: Z6lXHBKOyd.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/16 22:46:35.441754 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
BYE sip:1000@//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK48ef4999;rport.
Max-Forwards: 70.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 103 BYE.
User-Agent: Asterisk PBX 11.8.1.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.

#
U 2014/04/16 22:46:35.474403 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK48ef4999;rport.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 103 BYE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.

exit
21 received, 0 dropped

你知道为什么当设备在LTE(4G)网络上时它无法发出声音吗?

1 个答案:

答案 0 :(得分:0)

我在sip.conf和用户的conf中更改了以下代码。

canreinvite = yes 

这一切都很好。但是,它通过Asterisk Server传送声音,这意味着服务器必须处理语音流量。