I changed below code in sip.conf and user's conf.
canreinvite = yes
It all worked fine. However, it delivers sound through Asterisk Server, which means the server has to take care of voice traffic.
Question
I've installed Asterisk 11, and two wifi phones are fine to talk through asterisk server. However, a wifi phone and LTE(4G) phone can't deliver sounds.
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
Sip settings
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)
sip logs When I look at sip logs, it looks like fine. I just see one more "invite" from server to wifi-phone.
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
Do you see why it failed to deliver sound when a device is on LTE(4G) network?
Solution
I changed below code in sip.conf and user's conf.
canreinvite = yes
It all worked fine. However, it delivers sound through Asterisk Server, which means the server has to take care of voice traffic.