Freeswitch stops rtp streaming after 2nd invite sent

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Calvin Lee

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Jul 12, 2022
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Hi,

I have a voice gateway (hardware, in LAN), registering to FusionPBX running in remote cloud with public IP. The voice gateway has two fxs ports acting as two callcenter agents.

The issue is: when 1st call comes in, FusionPBX can successfully manage a session with an agent (22060001); but when the 2nd call comes in, the 1st call would hear nothing from agent side. With captured data I can see Freeswitch would stop streaming rtp packets to the 1st agent after inviting the 2nd agent (22060002). Can't see any error message within Freeswitch log or voice gateway log though.

Any help would be appreciated!

Notes:
1、The FusionPBX instance works fine with software voip clients (MicroSIP etc);
2、22060001 and 22060002 register to FusionPBX using a same IP & port, since they are configured on the same voice gateway HW box.

FusionPBX IP address / domain name: 121.89.x.y / pbx.domain.com
voice gateway IP address: 192.168.1.81

Below are log messages captured within voice gateway:
------------------------------------------------------------------------
[07/11 17:36:37.690914]SIP:recv[121.89.x.y:5060] len[1289]
INVITE sip:22060001@192.168.1.81:9887 SIP/2.0
Via: SIP/2.0/UDP 121.89.x.y;rport;branch=z9hG4bKtrKpmSj6arZ6Q
Route: <sip:22060001@192.168.2.3:9887>
Max-Forwards: 70
From: "01#100" <sip:100@pbx.domain.com>;tag=3aXyp8Fm37X8j
To: <sip:22060001@192.168.1.81:9887>
Call-ID: cd575bbd-7b9f-123b-52a2-00163e05f880
CSeq: 54212306 INVITE
Contact: <sip:mod_sofia@121.89.x.y:5060>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, 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
Session-Expires: 120;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
caller_destination: 02423221320
X-FS-Support: update_display,send_info
Remote-Party-ID: "01#100" <sip:100@pbx.domain.com>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1657500155 1657500156 IN IP4 121.89.x.y
s=FreeSWITCH
c=IN IP4 121.89.x.y
t=0 0
m=audio 32042 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

[07/11 17:36:37.748911]SIP:send[121.89.x.y:5060] len[306]
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKtrKpmSj6arZ6Q
To: <sip:22060001@192.168.1.81:9887>;tag=16575321971657530684-1
From: "01#100"<sip:100@pbx.domain.com>;tag=3aXyp8Fm37X8j
Call-ID: cd575bbd-7b9f-123b-52a2-00163e05f880
CSeq: 54212306 INVITE
Content-Length: 0


[07/11 17:36:37.752098]LINE-22060001(9) incoming cd575bbd-7b9f-123b-52a2-00163e05f880
[07/11 17:36:37.773398]SIP:send[121.89.x.y:5060] len[350]
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKtrKpmSj6arZ6Q
To: <sip:22060001@192.168.1.81:9887>;tag=16575321971657530684-1
From: "01#100"<sip:100@pbx.domain.com>;tag=3aXyp8Fm37X8j
Call-ID: cd575bbd-7b9f-123b-52a2-00163e05f880
CSeq: 54212306 INVITE
Contact: <sip:22060001@192.168.1.81:9887>
Content-Length: 0


[07/11 17:36:45.722398]LINE-22060001(9) offhook
[07/11 17:36:45.754549]SIP:send[121.89.x.y:5060] len[588]
SIP/2.0 200 OK
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKtrKpmSj6arZ6Q
To: <sip:22060001@192.168.1.81:9887>;tag=16575321971657530684-1
From: "01#100"<sip:100@pbx.domain.com>;tag=3aXyp8Fm37X8j
Call-ID: cd575bbd-7b9f-123b-52a2-00163e05f880
CSeq: 54212306 INVITE
Contact: <sip:22060001@192.168.1.81:9887>
Content-Type: application/sdp
Content-Length: 210

v=0
o=- 1657532205 742493 IN IP4 192.168.1.81
s=-
c=IN IP4 192.168.1.81
t=0 0
m=audio 10022 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

[07/11 17:36:45.796542]SIP:recv[121.89.x.y:5060] len[386]
ACK sip:22060001@192.168.1.81:9887 SIP/2.0
Via: SIP/2.0/UDP 121.89.x.y;rport;branch=z9hG4bKU1cFpm3970NSK
Max-Forwards: 70
From: "01#100" <sip:100@pbx.domain.com>;tag=3aXyp8Fm37X8j
To: <sip:22060001@192.168.1.81:9887>;tag=16575321971657530684-1
Call-ID: cd575bbd-7b9f-123b-52a2-00163e05f880
CSeq: 54212306 ACK
Contact: <sip:mod_sofia@121.89.x.y:5060>
Content-Length: 0


[07/11 17:36:58.300798]SIP:recv[121.89.x.y:5060] len[1289]
INVITE sip:22060002@192.168.1.81:9887 SIP/2.0
Via: SIP/2.0/UDP 121.89.x.y;rport;branch=z9hG4bKva67QFmD59BcF
Route: <sip:22060002@192.168.2.3:9887>
Max-Forwards: 70
From: "01#101" <sip:101@pbx.domain.com>;tag=r9r0m2cFa0t7r
To: <sip:22060002@192.168.1.81:9887>
Call-ID: d9a008a0-7b9f-123b-52a2-00163e05f880
CSeq: 54212317 INVITE
Contact: <sip:mod_sofia@121.89.x.y:5060>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, 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
Session-Expires: 120;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
caller_destination: 02423221320
X-FS-Support: update_display,send_info
Remote-Party-ID: "01#101" <sip:101@pbx.domain.com>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1657509748 1657509749 IN IP4 121.89.x.y
s=FreeSWITCH
c=IN IP4 121.89.x.y
t=0 0
m=audio 22470 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

[07/11 17:36:58.361561]SIP:send[121.89.x.y:5060] len[306]
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKva67QFmD59BcF
To: <sip:22060002@192.168.1.81:9887>;tag=16575322181657530686-1
From: "01#101"<sip:101@pbx.domain.com>;tag=r9r0m2cFa0t7r
Call-ID: d9a008a0-7b9f-123b-52a2-00163e05f880
CSeq: 54212317 INVITE
Content-Length: 0


[07/11 17:36:58.362292]LINE-22060002(10) incoming d9a008a0-7b9f-123b-52a2-00163e05f880
[07/11 17:36:58.395376]SIP:send[121.89.x.y:5060] len[350]
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKva67QFmD59BcF
To: <sip:22060002@192.168.1.81:9887>;tag=16575322181657530686-1
From: "01#101"<sip:101@pbx.domain.com>;tag=r9r0m2cFa0t7r
Call-ID: d9a008a0-7b9f-123b-52a2-00163e05f880
CSeq: 54212317 INVITE
Contact: <sip:22060002@192.168.1.81:9887>
Content-Length: 0


[07/11 17:37:13.146955]SIP:recv[121.89.x.y:5060] len[408]
CANCEL sip:22060002@192.168.1.81:9887 SIP/2.0
Via: SIP/2.0/UDP 121.89.x.y;rport;branch=z9hG4bKva67QFmD59BcF
Route: <sip:22060002@192.168.2.3:9887>
Max-Forwards: 70
From: "01#101" <sip:101@pbx.domain.com>;tag=r9r0m2cFa0t7r
To: <sip:22060002@192.168.1.81:9887>
Call-ID: d9a008a0-7b9f-123b-52a2-00163e05f880
CSeq: 54212317 CANCEL
Reason: SIP;cause=487;text="ORIGINATOR_CANCEL"
Content-Length: 0


[07/11 17:37:13.182934]SIP:send[121.89.x.y:5060] len[318]
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKva67QFmD59BcF
To: <sip:22060002@192.168.1.81:9887>;tag=16575322181657530686-1
From: "01#101"<sip:101@pbx.domain.com>;tag=r9r0m2cFa0t7r
Call-ID: d9a008a0-7b9f-123b-52a2-00163e05f880
CSeq: 54212317 INVITE
Content-Length: 0


[07/11 17:37:13.187486]SIP:send[121.89.x.y:5060] len[289]
SIP/2.0 200 OK
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKva67QFmD59BcF
To: <sip:22060002@192.168.1.81:9887>;tag=db9e1369
From: "01#101" <sip:101@pbx.domain.com>;tag=r9r0m2cFa0t7r
Call-ID: d9a008a0-7b9f-123b-52a2-00163e05f880
CSeq: 54212317 CANCEL
Content-Length: 0


[07/11 17:37:13.226813]SIP:recv[121.89.x.y:5060] len[381]
ACK sip:22060002@192.168.1.81:9887 SIP/2.0
Via: SIP/2.0/UDP 121.89.x.y;rport;branch=z9hG4bKva67QFmD59BcF
Route: <sip:22060002@192.168.2.3:9887>
Max-Forwards: 70
From: "01#101" <sip:101@pbx.domain.com>;tag=r9r0m2cFa0t7r
To: <sip:22060002@192.168.1.81:9887>;tag=16575322181657530686-1
Call-ID: d9a008a0-7b9f-123b-52a2-00163e05f880
CSeq: 54212317 ACK
Content-Length: 0


[07/11 17:37:16.588539]SIP:recv[121.89.x.y:5060] len[556]
BYE sip:22060001@192.168.1.81:9887 SIP/2.0
Via: SIP/2.0/UDP 121.89.x.y;rport;branch=z9hG4bKXKZ0Sa5g2j2ya
Max-Forwards: 70
From: "01#100" <sip:100@pbx.domain.com>;tag=3aXyp8Fm37X8j
To: <sip:22060001@192.168.1.81:9887>;tag=16575321971657530684-1
Call-ID: cd575bbd-7b9f-123b-52a2-00163e05f880
CSeq: 54212307 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0


[07/11 17:37:16.617215]SIP:send[121.89.x.y:5060] len[299]
SIP/2.0 200 OK
Via: SIP/2.0/UDP 121.89.x.y;rport=5060;branch=z9hG4bKXKZ0Sa5g2j2ya
To: <sip:22060001@192.168.1.81:9887>;tag=16575321971657530684-1
From: "01#100"<sip:100@pbx.domain.com>;tag=3aXyp8Fm37X8j
Call-ID: cd575bbd-7b9f-123b-52a2-00163e05f880
CSeq: 54212307 BYE
Content-Length: 0


[07/11 17:37:20.092652]LINE-22060001(9) onhook
 

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DigitalDaz

DigitalDaz

Administrator
Staff member
Sep 29, 2016
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Look again at the freeswitch logs and maybe do a: sofia global siptrace on

The Freeswitch appears to be hanging up the other call, I can see a BYE from it.
 
C

Calvin Lee

New Member
Jul 12, 2022
2
0
1
48
DigitalDaz said:
Look again at the freeswitch logs and maybe do a: sofia global siptrace on

The Freeswitch appears to be hanging up the other call, I can see a BYE from it.
Click to expand...
Thank you sir. Actually the bye msg at 17:37:16.588539 was caused by caller hang-up. As can seen from the snapshot image, rtp streaming stopped after 17:36:58.
 
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