[Freeswitch-users] Oneway audio issues in freeswitch

Jai Rangi jprangi at gmail.com
Thu Feb 11 06:45:30 MSK 2016 

We have been looking at that all day, but cant figure out the issue. Funny
thing is that its happening only when GS Originate the call. May be we are
over looking something. Here are two call example. IPs are modified for
security.

FreeSWITCH (Version 1.6.6  64bit) is ready
freeswitch at internal> sofia_contact 1276 at domain.company.com
sofia/internal/sip:1276 at 192.168.1.168:12113
;fs_nat=yes;fs_path=sip%3A1276%4068.5.194.163%3A12113
freeswitch at internal> sofia_contact 142 at domain.company.com
sofia/internal/sip:142 at 172.16.42.13:11852
;fs_nat=yes;fs_path=sip%3A142%4074.67.200.39%3A33812


142 calls 1276 (1276 does not hear anything) (Seems freeswitch not handling
nat properly) On TCP dump, I can see free switching receiving the RTP
packet, but trying to deliver to local IP for 1276.
1276 calls 142 (All good both parties can hear)
142 call PSTN number (All good)
1272 call PSTN number (All good)

Same configuration, same dialplan works just fine with 1.6.2 and 1.4.18.
1.6.2 had intermittent DTMF issue, we upgraded to 1.6.5, found this one way
audio, upgraded to 1.6.6.
We have narrowed it down to Grand Stream Softphone and Grad Stream IP
phones.
Here is trace for Both calls.

######################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
INVITE sip:1276 at domain.company.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 INVITE.
Contact: "142" <sip:142 at 172.16.42.13:11852>.
Max-Forwards: 70.
User-Agent: Grandstream Wave/IOS 1.0.19.
Privacy: none.
P-Preferred-Identity: "142" <sip:142 at domain.company.com:5060>.
Supported: replaces, path, timer, eventlist.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length:   235.
.
v=0.
o=142 8000 8000 IN IP4 172.16.42.13.
s=SIP Call.
c=IN IP4 172.16.42.13.
t=0 0.
m=audio 50476 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=Kt2jU0QN57N8F.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 INVITE.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Proxy-Authenticate: Digest realm="domain.company.com",
nonce="5eb04922-17e3-426e-9ddf-4333692102b5", algorithm=MD5, qop="auth".
Content-Length: 0.
.

##
U 74.67.200.39:33812 -> 222.222.222.222:5060
ACK sip:1276 at domain.company.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=Kt2jU0QN57N8F.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 140 ACK.
Content-Length: 0.
.

#
U 74.67.200.39:33812 -> 222.222.222.222:5060
INVITE sip:1276 at domain.company.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
Contact: "142" <sip:142 at 172.16.42.13:11852>.
Proxy-Authorization: Digest username="xxxxx", realm="domain.company.com",
nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri="
sip:1276 at domain.company.com:5060",
response="d309cb76b83042023f6794835ad60a89", algorithm=MD5,
cnonce="15121380", qop=auth, nc=00000003.
Max-Forwards: 70.
User-Agent: Grandstream Wave/IOS 1.0.19.
Privacy: none.
P-Preferred-Identity: "142" <sip:142 at domain.company.com:5060>.
Supported: replaces, path, timer, eventlist.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length:   235.
.
v=0.
o=142 8000 8000 IN IP4 172.16.42.13.
s=SIP Call.
c=IN IP4 172.16.42.13.
t=0 0.
m=audio 50476 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
INVITE sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKX5H9FDZX1eyNN.
Route: <sip:1276 at 68.5.194.163:12113>.
Max-Forwards: 68.
From: "Softphone" <sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: <sip:1276 at 192.168.1.168:12113>.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Contact: <sip:142 at 222.222.222.222:5060>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 270.
X-FS-Support: update_display,send_info.
.
v=0.
o=FreeSWITCH 1455142324 1455142325 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 17642 RTP/AVP 0 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
#
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 100 Trying.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.
Content-Length: 0.
.

###
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 180 Ringing.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 222.
P-Asserted-Identity: "1276" <sip:1276 at domain.company.com>.
.
v=0.
o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 27910 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

##############
##
#######
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 200 OK.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 INVITE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.
Supported: timer,replaces.
Content-Type: application/sdp.
Content-Length: 207.
.
v=0.
o=1276 87748 1 IN IP4 192.168.1.168.
s=-.
c=IN IP4 192.168.1.168.
t=0 0.
m=audio 8000 RTP/AVP 0 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:96 telephone-event/8000.
a=ptime:20.
a=rtpmap:96 telephone-event/8000.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
ACK sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKZQ4tK304U0aUc.
Max-Forwards: 70.
From: "Softphone" <sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: <sip:1276 at 192.168.1.168:12113>;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243919 ACK.
Contact: <sip:142 at 222.222.222.222:5060>.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 222.
P-Asserted-Identity: "Outbound Call" <sip:1276 at domain.company.com>.
.
v=0.
o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 27910 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

##
U 74.67.200.39:33812 -> 222.222.222.222:5060
ACK sip:1276 at 222.222.222.222:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1657841580;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 141 ACK.
Contact: <sip:142 at 172.16.42.13:11852>.
Proxy-Authorization: Digest username="xxxx", realm="domain.company.com",
nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri="
sip:1276 at domain.company.com:5060",
response="d309cb76b83042023f6794835ad60a89", algorithm=MD5,
cnonce="15121380", qop=auth, nc=00000003.
Max-Forwards: 70.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

###############################################################
####
#

#
#########
################################################################################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
BYE sip:1276 at 222.222.222.222:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK326133779;rport.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 142 BYE.
Contact: <sip:142 at 172.16.42.13:11852>.
Max-Forwards: 70.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK326133779;rport=33812;received=74.67.200.39.
From: "142" <sip:142 at domain.company.com:5060>;tag=479799221.
To: <sip:1276 at domain.company.com:5060>;tag=m3UBXU8r2gcUB.
Call-ID: 152039027-11852-15 at BHC.BG.EC.BD.
CSeq: 142 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Content-Length: 0.
.

###
U 222.222.222.222:5060 -> 68.5.194.163:12113
BYE sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK2jg5rmKFKUDKF.
Max-Forwards: 70.
From: "Softphone" <sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: <sip:1276 at 192.168.1.168:12113>;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243920 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

###
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 200 OK.
From: Softphone<sip:142 at domain.company.com>;tag=NcN4ypSvZS2DQ.
To: sip:1276 at 192.168.1.168:12113;tag=ReKf2-xIWCt.
Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243920 BYE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK2jg5rmKFKUDKF.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.
.

#######################################################
#########
#

Successful Call with 2 way audio :1276 calls 142

#
#
#
##################################
U 68.5.194.163:12113 -> 222.222.222.222:5060
INVITE sip:142 at domain.company.com SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 INVITE.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
Max-Forwards: 70.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.
Supported: timer,replaces.
Content-Type: application/sdp.
Content-Length: 257.
.
v=0.
o=1276 87384 1 IN IP4 192.168.1.168.
s=-.
c=IN IP4 192.168.1.168.
t=0 0.
m=audio 8002 RTP/AVP 0 18 8 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:96 telephone-event/8000.
a=ptime:20.
a=rtpmap:96 telephone-event/8000.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>;tag=SjctQ6jcUQD5a.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 INVITE.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Proxy-Authenticate: Digest realm="domain.company.com",
nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb", algorithm=MD5, qop="auth".
Content-Length: 0.
.

#
U 68.5.194.163:12113 -> 222.222.222.222:5060
ACK sip:142 at domain.company.com SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com;tag=SjctQ6jcUQD5a.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 176 ACK.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH.
Content-Length: 0.
.

#
U 68.5.194.163:12113 -> 222.222.222.222:5060
INVITE sip:142 at domain.company.com SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
Max-Forwards: 70.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER.
Supported: timer,replaces.
Proxy-Authorization: Digest username="yyyy",realm="domain.company.com",uri="
sip:142 at domain.company.com
",response="a87ca168b1869994a6b4782df7bffe99",algorithm=MD5,nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb",qop=auth,cnonce="00017214",nc=00000001.
Content-Type: application/sdp.
Content-Length: 257.
.
v=0.
o=1276 87384 1 IN IP4 192.168.1.168.
s=-.
c=IN IP4 192.168.1.168.
t=0 0.
m=audio 8002 RTP/AVP 0 18 8 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:96 telephone-event/8000.
a=ptime:20.
a=rtpmap:96 telephone-event/8000.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
User-Agent: VOIPGATEWAY.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
INVITE sip:142 at 172.16.42.13:11852 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD.
Route: <sip:142 at 74.67.200.39:33812>.
Max-Forwards: 68.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 270.
X-FS-Support: update_display,send_info.
.
v=0.
o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 18346 RTP/AVP 0 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#

U 222.222.222.222:5060 -> 74.67.200.39:33812
INVITE sip:142 at 172.16.42.13:11852 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD.
Route: <sip:142 at 74.67.200.39:33812>.
Max-Forwards: 68.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:1276 at 222.222.222.222:5060>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 270.
X-FS-Support: update_display,send_info.
.
v=0.
o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 18346 RTP/AVP 0 8 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

##
U 74.67.200.39:33812 -> 222.222.222.222:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 222.222.222.222:5060
;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Supported: replaces, path, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 74.67.200.39:33812 -> 222.222.222.222:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 222.222.222.222:5060
;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:142 at 172.16.42.13:11852>.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow-Events: talk, hold.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>;tag=tU5jS13Fr03Qp.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
Contact: <sip:142 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 219.
Remote-Party-ID: "142" <sip:142 at domain.company.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 19096 RTP/AVP 0 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:96 telephone-event/8000.
a=fmtp:96 0-16.
a=ptime:20.

##################################################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 222.222.222.222:5060
;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 INVITE.
Contact: <sip:142 at 172.16.42.13:11852>.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Type: application/sdp.
Content-Length:   235.
.
v=0.
o=142 8000 8000 IN IP4 172.16.42.13.
s=SIP Call.
c=IN IP4 172.16.42.13.
t=0 0.
m=audio 26390 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
ACK sip:142 at 172.16.42.13:11852 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKB4K487aFSBpUQ.
Max-Forwards: 70.
From: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
To: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243932 ACK.
Contact: <sip:1276 at 222.222.222.222:5060>.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.168:12113
;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163.
From: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: <sip:142 at domain.company.com>;tag=tU5jS13Fr03Qp.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 INVITE.
Contact: <sip:142 at 222.222.222.222:5060;transport=udp>.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 219.
Remote-Party-ID: "Outbound Call" <sip:142 at domain.company.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222.
s=FreeSWITCH.
c=IN IP4 222.222.222.222.
t=0 0.
m=audio 19096 RTP/AVP 0 96.
a=rtpmap:0 PCMU/8000.
a=rtpmap:96 telephone-event/8000.
a=fmtp:96 0-16.
a=ptime:20.

#
U 68.5.194.163:12113 -> 222.222.222.222:5060
ACK sip:142 at 222.222.222.222:5060 SIP/2.0.
From: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
To: sip:142 at domain.company.com;tag=tU5jS13Fr03Qp.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 177 ACK.
Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-VwM5aez*.
Contact: ATAPHONE<sip:1276 at 192.168.1.168:12113>.
Max-Forwards: 70.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.
.

###########################################################################################
##############################
##############################################################################################################################################################
U 74.67.200.39:33812 -> 222.222.222.222:5060
BYE sip:1276 at 222.222.222.222:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK364522226;rport.
From: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
To: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243933 BYE.
Contact: <sip:142 at 172.16.42.13:11852>.
Max-Forwards: 70.
Supported: replaces, path, timer, eventlist.
User-Agent: Grandstream Wave/IOS 1.0.19.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER,
UPDATE, MESSAGE.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 74.67.200.39:33812
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.42.13:11852
;branch=z9hG4bK364522226;rport=33812;received=74.67.200.39.
From: <sip:142 at 172.16.42.13:11852>;tag=1713531477.
To: "ATA" <sip:1276 at domain.company.com>;tag=U4yBUvmKN9Saj.
Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c.
CSeq: 87243933 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Content-Length: 0.
.

#
U 222.222.222.222:5060 -> 68.5.194.163:12113
BYE sip:1276 at 192.168.1.168:12113 SIP/2.0.
Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK89gKvrKN5DvSQ.
Max-Forwards: 70.
From: <sip:142 at domain.company.com>;tag=tU5jS13Fr03Qp.
To: ATAPHONE <sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 87243943 BYE.
User-Agent: VOIPGATEWAY.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.

##
U 68.5.194.163:12113 -> 222.222.222.222:5060
SIP/2.0 200 OK.
From: sip:142 at domain.company.com;tag=tU5jS13Fr03Qp.
To: ATAPHONE<sip:1276 at domain.company.com>;tag=VxLf2-dmKe50.
Call-ID: d44p80-lYfd5f2 at domain.company.com.
CSeq: 87243943 BYE.
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK89gKvrKN5DvSQ.
User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>.
Content-Length: 0.





On Wed, Feb 10, 2016 at 5:54 PM, Ítalo Rossi <italo at freeswitch.org> wrote:

> You need to look at the sip signaling to see what's going on
>
> On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v at gmail.com>
> wrote:
>
>> Hello All
>>
>> We are getting one way audio issues with some softphones and grandstream
>> phones behind nat registerd to our freeswitch server.
>>
>> Here is scenario:
>> Grandstream call any extensions (one way audio)
>> Any extension call Grandstream ( Audio works just fine)
>>
>> We have tried multiple softphones and the result is same.
>>
>> Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF
>> issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started
>> with an upgrade to freeswitch.
>>
>> Any help or hint will be much appreciated.
>>
>> Thank you,
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ítalo Rossi
> italo at freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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