No Audio in Inbound Calls!

xserver · August 29, 2022, 5:11pm

I have installed ASTPP v5.0 Edition and currently facing issues with No Audio on Inbound calls.
During Inbound call , the Caller can hear Callee , but Callee cannot hear Caller Audio.
During Outbound both side audio works fine.

Here are my IP and call trace.

SIP Server IP : 10.229.56.20
ASTPP SIP Provider Side IP : 10.229.55.217
ASTPP LAN IP : 10.128.0.1
ASTPP Public Management IP : 103.103.103.138
DID Extension : 01234228425
SIP Phone IP : 103.103.103.103
Inbound Caller : +911234567899

Attached is the call trace of both leg

Leg A

2022/08/29 22:12:40.287097 10.229.56.20:5060 -> 10.229.55.217:5060
INVITE sip:01234228425@10.229.55.217:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bK5nusgv1010paonfj02o0.1
Call-ID: 7gyibfg9ebf951y6gccflifb59jl17ji@UAC
From: <sip:+911234567899@10.229.56.20;user=phone>;tag=1i6ib988-CC-1004
To: <sip:01234228425@10.229.55.217;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:+911234567899@192.168.73.4;user=phone>
Max-Forwards: 68
Contact: <sip:+911234567899@10.229.56.20:5060;transport=udp>
Min-SE: 90
Session-Expires: 1800
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V600R013
Supported: 100rel,timer
Content-Length: 208
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 2071952591 2071952592 IN IP4 10.229.56.20
s=Sip Call
c=IN IP4 10.229.56.20
t=0 0
m=audio 61156 RTP/AVP 8 99
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15

2022/08/29 22:12:40.289505 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bK5nusgv1010paonfj02o0.1
From: <sip:+911234567899@10.229.56.20;user=phone>;tag=1i6ib988-CC-1004
To: <sip:01234228425@10.229.55.217;user=phone>
Call-ID: 7gyibfg9ebf951y6gccflifb59jl17ji@UAC
CSeq: 1 INVITE
User-Agent: ASTPP
Content-Length: 0


2022/08/29 22:12:40.492249 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bK5nusgv1010paonfj02o0.1
From: <sip:+911234567899@10.229.56.20;user=phone>;tag=1i6ib988-CC-1004
To: <sip:01234228425@10.229.55.217;user=phone>;tag=FBy1UFprZNrQD
Call-ID: 7gyibfg9ebf951y6gccflifb59jl17ji@UAC
CSeq: 1 INVITE
Contact: <sip:01234228425@10.229.55.217:5060;transport=udp>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
P-Asserted-Identity: "Outbound Call" <sip:01234228425@10.229.55.217>


2022/08/29 22:12:42.118918 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bK5nusgv1010paonfj02o0.1
From: <sip:+911234567899@10.229.56.20;user=phone>;tag=1i6ib988-CC-1004
To: <sip:01234228425@10.229.55.217;user=phone>;tag=FBy1UFprZNrQD
Call-ID: 7gyibfg9ebf951y6gccflifb59jl17ji@UAC
CSeq: 1 INVITE
Contact: <sip:01234228425@10.229.55.217:5060;transport=udp>

Leg B

2022/08/29 22:12:40.371492 10.128.0.1:5060 -> 103.103.103.103:59254
INVITE sip:01234228425@103.103.103.103:59254;rinstance=681c928a3c366dad SIP/2.0
Via: SIP/2.0/UDP 10.229.55.217;rport;branch=z9hG4bK8KFy0Nvg4353g
Max-Forwards: 67
From: "+911234567899" <sip:+911234567899@103.103.103.138>;tag=HXZgKU491UmXK
To: <sip:01234228425@103.103.103.103:59254;rinstance=681c928a3c366dad>
Call-ID: 70216083-a25c-123b-a2bf-000c29ed313a
CSeq: 56341888 INVITE
Contact: <sip:mod_sofia@10.229.55.217:5060>
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, 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
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 222
P-Asserted-Identity: "+911234567899" <sip:+911234567899@103.103.103.138>

v=0
o=FreeSWITCH 1661766480 1661766481 IN IP4 10.229.55.217
s=FreeSWITCH
c=IN IP4 10.229.55.217
t=0 0
m=audio 24880 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2022/08/29 22:12:40.479982 103.103.103.103:59254 -> 10.128.0.1:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.229.55.217;rport=5060;branch=z9hG4bK8KFy0Nvg4353g;received=10.128.0.1
Contact: <sip:01234228425@103.103.103.103:59254;rinstance=681c928a3c366dad>
To: <sip:01234228425@103.103.103.103:59254;rinstance=681c928a3c366dad>;tag=820e2822
From: "+911234567899"<sip:+911234567899@103.103.103.138>;tag=HXZgKU491UmXK
Call-ID: 70216083-a25c-123b-a2bf-000c29ed313a
CSeq: 56341888 INVITE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0


2022/08/29 22:12:42.092523 103.103.103.103:59254 -> 10.128.0.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.229.55.217;rport=5060;branch=z9hG4bK8KFy0Nvg4353g;received=10.128.0.1
Contact: <sip:01234228425@103.103.103.103:59254;rinstance=681c928a3c366dad>
To: <sip:01234228425@103.103.103.103:59254;rinstance=681c928a3c366dad>;tag=820e2822
From: "+911234567899"<sip:+911234567899@103.103.103.138>;tag=HXZgKU491UmXK
Call-ID: 70216083-a25c-123b-a2bf-000c29ed313a
CSeq: 56341888 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 188

v=0
o=- 0 2 IN IP4 100.65.106.181
s=CounterPath eyeBeam 1.5
c=IN IP4 100.65.106.181
t=0 0
m=audio 19666 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

I would appreciate any pointers to what should be done in order to resolve the issue of 1 side audio.

KNERD · August 29, 2022, 5:29pm

It is clear the ASTPP server is behind a router. This more than likely a NAT/firewall issue.

xserver · August 29, 2022, 5:31pm

There is no NAT involved in the current scenario. SIP devices are connected to SIP server in routing mode and SIP Provider is connected via routing.

I have already checked twice for the NAT issue but there is no NAT involved.

xserver · August 30, 2022, 10:22am

Sorry Yes, there was a NAT issue which is now resolved. But still the problem persists . Outbound calls are working fine but inbound calls have one side audio ( Caller voice is not audible to Callee ).

I have created 2 profiles ,

1 - Internal for local SIP devices running on 10.128.0.1:5060
2 - External for Gateway SIP running on 10.229.55.217:5060

As per freeswitch , we do not need to create external profile on port 5060 but if i use 5070 / 5080 , inbound calls do not reach freeswitch.

Posting the new call trace logs again for reference.

SIP Server IP : 10.229.56.20
ASTPP Provider Side IP : 10.229.55.217
ASTPP LAN IP : 10.128.0.1
SIP Phone IP : 10.131.15.234
Inbound Caller : +819144659796
SIP / DID Destination : 32171228425

Leg A Call Trace

2022/08/30 15:41:28.412271 10.229.56.20:5060 -> 10.229.55.217:5060
INVITE sip:32171228425@10.229.55.217:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bKvcdc5g20185cpel37rf0.1
Call-ID: fb5ikeg7kkl7ebe1ylk7yj68gg6l1116@UAC
From: <sip:+819144659796@10.229.56.20;user=phone>;tag=iiy6gf7y-CC-1001
To: <sip:32171228425@10.229.55.217;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:+819144659796@192.168.73.4;user=phone>
Max-Forwards: 68
Contact: <sip:+819144659796@10.229.56.20:5060;transport=udp>
Min-SE: 90
Session-Expires: 1800
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V600R013
Supported: 100rel,timer
Content-Length: 208
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 1984292210 1984292211 IN IP4 10.229.56.20
s=Sip Call
c=IN IP4 10.229.56.20
t=0 0
m=audio 15852 RTP/AVP 8 99
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15

2022/08/30 15:41:28.414720 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bKvcdc5g20185cpel37rf0.1
From: <sip:+819144659796@10.229.56.20;user=phone>;tag=iiy6gf7y-CC-1001
To: <sip:32171228425@10.229.55.217;user=phone>
Call-ID: fb5ikeg7kkl7ebe1ylk7yj68gg6l1116@UAC
CSeq: 1 INVITE
User-Agent: ASTPP
Content-Length: 0


2022/08/30 15:41:28.698749 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bKvcdc5g20185cpel37rf0.1
From: <sip:+819144659796@10.229.56.20;user=phone>;tag=iiy6gf7y-CC-1001
To: <sip:32171228425@10.229.55.217;user=phone>;tag=234HDc2FZa44p
Call-ID: fb5ikeg7kkl7ebe1ylk7yj68gg6l1116@UAC
CSeq: 1 INVITE
Contact: <sip:32171228425@10.229.55.217:5060;transport=udp>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
P-Asserted-Identity: "Outbound Call" <sip:32171228425@10.229.55.217>


2022/08/30 15:41:30.813787 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bKvcdc5g20185cpel37rf0.1
From: <sip:+819144659796@10.229.56.20;user=phone>;tag=iiy6gf7y-CC-1001
To: <sip:32171228425@10.229.55.217;user=phone>;tag=234HDc2FZa44p
Call-ID: fb5ikeg7kkl7ebe1ylk7yj68gg6l1116@UAC
CSeq: 1 INVITE
Contact: <sip:32171228425@10.229.55.217:5060;transport=udp>

Leg B Call Trace

2022/08/30 15:41:28.477617 10.128.0.1:5060 -> 10.131.15.234:50521
INVITE sip:32171228425@10.131.15.234:50521;transport=TCP;rinstance=37aec28af67741a2 SIP/2.0
Via: SIP/2.0/TCP 10.128.0.1;rport;branch=z9hG4bKr4Xc7y65pe5Zr
Max-Forwards: 67
From: "+819144659796" <sip:+819144659796@10.128.0.1>;tag=6F8Srv9DXNNrj
To: <sip:32171228425@10.131.15.234:50521;transport=TCP;rinstance=37aec28af67741a2>
Call-ID: f4346204-a2ee-123b-228e-000c29ed313a
CSeq: 56373352 INVITE
Contact: <sip:mod_sofia@10.128.0.1:5060;transport=tcp>
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, 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
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 216
P-Asserted-Identity: "+819144659796" <sip:+819144659796@10.128.0.1>

v=0
o=FreeSWITCH 1661836602 1661836603 IN IP4 10.128.0.1
s=FreeSWITCH
c=IN IP4 10.128.0.1
t=0 0
m=audio 17686 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2022/08/30 15:41:28.684043 10.131.15.234:50521 -> 10.128.0.1:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.128.0.1;rport=5060;branch=z9hG4bKr4Xc7y65pe5Zr
Contact: <sip:32171228425@10.131.15.234:50521;transport=TCP;rinstance=37aec28af67741a2>
To: <sip:32171228425@10.131.15.234:50521;transport=TCP;rinstance=37aec28af67741a2>;tag=3c6b6e63
From: "+819144659796"<sip:+819144659796@10.128.0.1>;tag=6F8Srv9DXNNrj
Call-ID: f4346204-a2ee-123b-228e-000c29ed313a
CSeq: 56373352 INVITE
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 0


2022/08/30 15:41:30.803674 10.131.15.234:50521 -> 10.128.0.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.128.0.1;rport=5060;branch=z9hG4bKr4Xc7y65pe5Zr
Contact: <sip:32171228425@10.131.15.234:50521;transport=TCP;rinstance=37aec28af67741a2>
To: <sip:32171228425@10.131.15.234:50521;transport=TCP;rinstance=37aec28af67741a2>;tag=3c6b6e63
From: "+819144659796"<sip:+819144659796@10.128.0.1>;tag=6F8Srv9DXNNrj
Call-ID: f4346204-a2ee-123b-228e-000c29ed313a
CSeq: 56373352 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 186

v=0
o=- 7 2 IN IP4 10.131.15.234
s=CounterPath eyeBeam 1.5
c=IN IP4 10.131.15.234
t=0 0
m=audio 14028 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

Any pointers about what might be the error will be highly appreciated.

harsh.inextrix · August 30, 2022, 11:51am

What is the value of ext-rtp-ip in Sip-Profile?

If it’s $${local_ip_v4} try to set ASTPP public IP and then try(Don’t forget to Reload Sip-Profile after changes).

xserver · August 30, 2022, 1:08pm

Hi Harsh,

ASPP Public IP is only for management , SIP service is running on 10.128.0.1:5060 ( separate dedicated ethernet port). SIP service is not exposed over public ip , it is only for local subnet users.

I have 2 SIP Profiles , Internal & External

Internal Profile is on 10.128.0.1:5060 which is facing towards SIP Phones in network.
External Profile is on 10.229.55.217:5060 which is facing towards Provider.

eval $${local_ip_v4} in fs_cli shows 10.128.0.1 , and is set in as ext-rtp-ip and ext-sip-ip in Internal Profile.

In External profile no ext-rtp-ip , ext-sip-ip attribute is present.

External Profile

freeswitch@zsip> sofia status profile VODA_EXT
=================================================================================================
Name                    VODA_EXT
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_VODA_EXT
Pres Hosts
Dialplan                XML
Context                 default
Challenge Realm         auto_to
RTP-IP                  10.229.55.217
Ext-RTP-IP              10.128.0.1
SIP-IP                  10.229.55.217
URL                     sip:mod_sofia@10.229.55.217:5060
BIND-URL                sip:mod_sofia@10.229.55.217:5060;transport=udp,tcp
HOLD-MUSIC              N/A
OUTBOUND-PROXY          N/A
CODECS IN               PCMU,PCMA,G729
CODECS OUT              PCMU,PCMA,G729
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
MAX-RECV-RPS            1000
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               2
FAILED-CALLS-OUT        1
REGISTRATIONS           0

Internal Profile

freeswitch@zsip> sofia status profile INTERNAL
=================================================================================================
Name                    INTERNAL
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_INTERNAL
Pres Hosts              10.128.0.1,10.128.0.1
Dialplan                XML
Context                 default
Challenge Realm         auto_from
RTP-IP                  10.128.0.1
Ext-RTP-IP              10.128.0.1
SIP-IP                  10.128.0.1
Ext-SIP-IP              10.128.0.1
URL                     sip:mod_sofia@10.128.0.1:5060
BIND-URL                sip:mod_sofia@10.128.0.1:5060;maddr=10.128.0.1;transport=udp,tcp
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               PCMU,PCMA,G729
CODECS OUT              PCMU,PCMA,G729
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
MAX-RECV-RPS            1000
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
CALLS-IN                4
FAILED-CALLS-IN         1
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           92

With above outbound calls are good on both side audio , but inbound calls 1 side audio.

harsh.inextrix · August 31, 2022, 8:15am

You can try to add a new param ext-rtp-ip with your RTP IP(In your case is 10.229.55.217 ) in your external profile. after add params I think your audio issue will solve.

xserver · September 1, 2022, 5:18am

Hi Harsh,

If my understanding is correct then ext-rtp-ip is needed only in NAT scenario. My setup is purely on routing and no NAT involved , is it necessary for me to use ext-rtp-ip ?

I have changed ext-rtp-ip but still no luck in inbound audio. Do let me know what logs you need to check the issue.

harsh.inextrix · September 1, 2022, 6:18am

I see in your log RTP comes on private IP that’s why I suggest you set public IP in ext-rtp-ip.

Share full Pcap file let me try to check in detail if I found anything other in logs.

xserver · September 1, 2022, 6:22am

HI @harsh.inextrix ,

The ASTPP box is completely on private IP with direct access to local sip users and providers.

ASTPP LAN IP : 10.128.0.1
ASTPP Provider Side IP : 10.229.55.217

freeswitch@zsip> sofia status profile ZAP_LOCAL_PROFILE
=================================================================================================
Name                    ZAP_LOCAL_PROFILE
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_ZAP_LOCAL_PROFILE
Pres Hosts              10.128.0.1,10.128.0.1
Dialplan                XML
Context                 default
Challenge Realm         auto_from
RTP-IP                  10.128.0.1 (LAN1 IP Connected to IP Phones)
Ext-RTP-IP              10.128.0.1
SIP-IP                  10.128.0.1
Ext-SIP-IP              10.128.0.1
URL                     sip:mod_sofia@10.128.0.1:5060
BIND-URL                sip:mod_sofia@10.128.0.1:5060;maddr=10.128.0.1;transport=udp,tcp
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               PCMU,PCMA,G729
CODECS OUT              PCMU,PCMA,G729
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
MAX-RECV-RPS            1000
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
CALLS-IN                6
FAILED-CALLS-IN         2
CALLS-OUT               1
FAILED-CALLS-OUT        0
REGISTRATIONS           2

freeswitch@zsip> sofia status profile EXTERNAL_PROFILE
=================================================================================================
Name                    VODA_EXT_PROFILE
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_EXTERNAL_PROFILE
Pres Hosts
Dialplan                XML
Context                 default
Challenge Realm         auto_to
RTP-IP                  10.229.55.217 (LAN2 IP Connected to SIP Provider)
Ext-RTP-IP              10.229.55.217
SIP-IP                  10.229.55.217
Ext-SIP-IP              10.229.55.217
URL                     sip:mod_sofia@10.229.55.217:5060
BIND-URL                sip:mod_sofia@10.229.55.217:5060;transport=udp,tcp
HOLD-MUSIC              N/A
OUTBOUND-PROXY          N/A
CODECS IN               PCMU,PCMA,G729
CODECS OUT              PCMU,PCMA,G729
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
MAX-RECV-RPS            1000
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
CALLS-IN                1
FAILED-CALLS-IN         0
CALLS-OUT               3
FAILED-CALLS-OUT        2
REGISTRATIONS           0

No Audio on Inbound calls , outbound calls working fine.

harsh.inextrix · September 1, 2022, 7:55am

Can you please share full pcap file for No Working(No Audio on Inbound calls) and working( outbound calls working fine) flow.

xserver · September 4, 2022, 11:42am

Posting the complete profile and call debug of working and non working.

Sofia Profiles

freeswitch@cc> sofia status profile Internal
=================================================================================================
Name                    Internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_Internal
Pres Hosts              100.101.102.103,100.101.102.103
Dialplan                XML
Context                 default
Challenge Realm         auto_from
RTP-IP                  100.101.102.103
Ext-RTP-IP              100.101.102.103
SIP-IP                  100.101.102.103
Ext-SIP-IP              100.101.102.103
URL                     sip:mod_sofia@100.101.102.103:5060
BIND-URL                sip:mod_sofia@100.101.102.103:5060;maddr=100.101.102.103;transport=udp,tcp
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               PCMA,PCMU
CODECS OUT              PCMA,PCMU
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
MAX-RECV-RPS            1000
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           true
CALLS-IN                2
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           1

freeswitch@cc> sofia status profile External
=================================================================================================
Name                    External
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_External
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  10.229.55.217
SIP-IP                  10.229.55.217
URL                     sip:mod_sofia@10.229.55.217:5060
BIND-URL                sip:mod_sofia@10.229.55.217:5060;transport=udp,tcp
HOLD-MUSIC              N/A
OUTBOUND-PROXY          N/A
CODECS IN               PCMU,PCMA,G729
CODECS OUT              PCMU,PCMA,G729
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
MAX-RECV-RPS            1000
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               1
FAILED-CALLS-OUT        0
REGISTRATIONS           0

Working Outgoing Calls Debug Log (SNGREP)

2022/09/04 16:56:29.804012 100.99.17.156:60642 -> 100.101.102.103:5060
INVITE sip:01234567890@cc.myserver.com SIP/2.0
Via: SIP/2.0/UDP 100.99.17.156:60642;branch=z9hG4bK-d8754z-c0787645753cae34-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4727985745@100.99.17.156:60642>
To: <sip:01234567890@cc.myserver.com>
From: "4727985745"<sip:4727985745@cc.myserver.com>;tag=412e2c22
Call-ID: Yzk5NGFmZjQ5MjBkYTNlOTA4ZDlkZTQ5Yzg2ZTk2NTc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 498

v=0
o=- 6 2 IN IP4 100.99.17.156
s=CounterPath Bria Professional
c=IN IP4 100.99.17.156
t=0 0
m=audio 25390 RTP/AVP 107 119 100 106 0 98 8 18 101
a=alt:1 1 : R1vMO3y6 LL/kJ8Vx 192.168.100.3 25390
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:5632259641714A73A0BDDEF8C78C3EC5

2022/09/04 16:56:29.806574 100.101.102.103:5060 -> 100.99.17.156:60642
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 100.99.17.156:60642;branch=z9hG4bK-d8754z-c0787645753cae34-1---d8754z-;rport=60642
From: "4727985745" <sip:4727985745@cc.myserver.com>;tag=412e2c22
To: <sip:01234567890@cc.myserver.com>;tag=Q4S9vFXQD052H
Call-ID: Yzk5NGFmZjQ5MjBkYTNlOTA4ZDlkZTQ5Yzg2ZTk2NTc.
CSeq: 1 INVITE
User-Agent: ASTPP
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="cc.myserver.com", nonce="bdf48834-6ee0-475b-912a-4f1667ed16e1", algorithm=MD5, qop="auth"
Content-Length: 0

2022/09/04 16:56:29.977443 10.229.55.217:5060 -> 10.229.56.20:5060
INVITE sip:01234567890@10.229.56.20 SIP/2.0
Via: SIP/2.0/UDP 10.229.55.217;rport;branch=z9hG4bKj0Xy9gKUUcmtF
Max-Forwards: 69
From: "09876543210" <sip:09876543210@10.229.55.217>;tag=tNSa9mUvjUFrc
To: <sip:01234567890@10.229.56.20>
Call-ID: 435f7d33-a6e7-123b-1f8a-000c299e1e72
CSeq: 56591602 INVITE
Contact: <sip:gw+VODA_DEL_GW@10.229.55.217:5060;transport=udp;gw=VODA_DEL_GW>
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248
X-FS-Support: update_display,send_info
Remote-Party-ID: "09876543210" <sip:09876543210@10.229.55.217>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1662269743 1662269744 IN IP4 10.229.55.217
s=FreeSWITCH
c=IN IP4 10.229.55.217
t=0 0
m=audio 21046 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2022/09/04 16:56:29.990088 10.229.56.20:5060 -> 10.229.55.217:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.229.55.217;received=10.229.55.217;branch=z9hG4bKj0Xy9gKUUcmtF;rport=5060
From: "09876543210" <sip:09876543210@10.229.55.217>;tag=tNSa9mUvjUFrc
To: <sip:01234567890@10.229.56.20>
Call-ID: 435f7d33-a6e7-123b-1f8a-000c299e1e72
CSeq: 56591602 INVITE
Content-Length: 0


2022/09/04 16:56:30.781169 10.229.56.20:5060 -> 10.229.55.217:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.229.55.217;received=10.229.55.217;branch=z9hG4bKj0Xy9gKUUcmtF;rport=5060
From: "09876543210" <sip:09876543210@10.229.55.217>;tag=tNSa9mUvjUFrc
To: <sip:01234567890@10.229.56.20>;tag=ellb5fgf-CC-1001
Call-ID: 435f7d33-a6e7-123b-1f8a-000c299e1e72
CSeq: 56591602 INVITE
Contact: <sip:01234567890@10.229.56.20:5060;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER

Non Working Inbound Call ( Caller - Callee No Sound , Callee - Caller Sound OK )

2022/09/04 17:01:54.703136 10.229.56.20:5060 -> 10.229.55.217:5060
INVITE sip:09876543210@10.229.55.217:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bKflv28c30480n70736ta0.1
Call-ID: li51ecgli915el67k9iy99gebcljkbk9@UAC
From: <sip:01234567890@10.229.56.20;user=phone>;tag=l8jj6i8k-CC-1003
To: <sip:09876543210@10.229.55.217;user=phone>
CSeq: 1 INVITE
P-Asserted-Identity: <sip:01234567890@192.168.73.4;user=phone>
Max-Forwards: 68
Contact: <sip:01234567890@10.229.56.20:5060;transport=udp>
Min-SE: 90
Session-Expires: 1800
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V600R013
Supported: 100rel,timer
Content-Length: 208
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 2099647963 2099647964 IN IP4 10.229.56.20
s=Sip Call
c=IN IP4 10.229.56.20
t=0 0
m=audio 26922 RTP/AVP 8 99
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15

2022/09/04 17:01:54.706422 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bKflv28c30480n70736ta0.1
From: <sip:01234567890@10.229.56.20;user=phone>;tag=l8jj6i8k-CC-1003
To: <sip:09876543210@10.229.55.217;user=phone>
Call-ID: li51ecgli915el67k9iy99gebcljkbk9@UAC
CSeq: 1 INVITE
User-Agent: ASTPP
Content-Length: 0


2022/09/04 17:01:55.159679 10.229.55.217:5060 -> 10.229.56.20:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.229.56.20:5060;branch=z9hG4bKflv28c30480n70736ta0.1
From: <sip:01234567890@10.229.56.20;user=phone>;tag=l8jj6i8k-CC-1003
To: <sip:09876543210@10.229.55.217;user=phone>;tag=618KyS56F6trQ
Call-ID: li51ecgli915el67k9iy99gebcljkbk9@UAC
CSeq: 1 INVITE
Contact: <sip:09876543210@10.229.55.217:5060;transport=udp>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY


2022/09/04 17:01:54.797223 100.101.102.103:5060 -> 100.99.17.156:60642
INVITE sip:4727985745@100.99.17.156:60642;rinstance=0037611f3898fbcf SIP/2.0
Via: SIP/2.0/UDP 100.101.102.103;rport;branch=z9hG4bKN1pg8mS6479BB
Max-Forwards: 67
From: "01234567890" <sip:01234567890@100.101.102.103>;tag=SpcU05yy7Hj8r
To: <sip:4727985745@100.99.17.156:60642;rinstance=0037611f3898fbcf>
Call-ID: 04fb0f8e-a6e8-123b-1f8a-000c299e1e72
CSeq: 56591765 INVITE
Contact: <sip:mod_sofia@100.101.102.103:5060>
User-Agent: ASTPP
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
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 222
P-Asserted-Identity: "01234567890" <sip:01234567890@100.101.102.103>

v=0
o=FreeSWITCH 1662263434 1662263435 IN IP4 100.101.102.103
s=FreeSWITCH
c=IN IP4 100.101.102.103
t=0 0
m=audio 27680 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2022/09/04 17:01:55.136546 100.99.17.156:60642 -> 100.101.102.103:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 100.101.102.103;rport=5060;branch=z9hG4bKN1pg8mS6479BB
To: <sip:4727985745@100.99.17.156:60642;rinstance=0037611f3898fbcf>
From: "01234567890" <sip:01234567890@100.101.102.103>;tag=SpcU05yy7Hj8r
Call-ID: 04fb0f8e-a6e8-123b-1f8a-000c299e1e72
CSeq: 56591765 INVITE
Content-Length: 0


2022/09/04 17:01:55.145214 100.99.17.156:60642 -> 100.101.102.103:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 100.101.102.103;rport=5060;branch=z9hG4bKN1pg8mS6479BB
Contact: <sip:4727985745@100.99.17.156:60642;rinstance=0037611f3898fbcf>
To: <sip:4727985745@100.99.17.156:60642;rinstance=0037611f3898fbcf>;tag=bf270640
From: "01234567890"<sip:01234567890@100.101.102.103>;tag=SpcU05yy7Hj8r
Call-ID: 04fb0f8e-a6e8-123b-1f8a-000c299e1e72
CSeq: 56591765 INVITE
User-Agent: Bria Professional release 2.4 stamp 49381
Content-Length: 0

xserver · September 4, 2022, 12:57pm

Posting fs_cli logs with siptrace ON

xserver · September 12, 2022, 5:54am

@harsh.inextrix

New update on the issue .

My SIP Trunk provider is Voafone , Inbound calls from Vodafone Network to ASTPP have both side audio.

Calls from other network to Vodafone SIP Trunk have 1 way audio issue.

Incoming Codec in both scenario is PCMA but it works on Vodafone network and doesn’t work from any other network.