SOLVED No Audio and call disconnects after 25 seconds.

[vchohan]

Hi Team,

We have freshly installed the incredible PBX, however, on calling b/w exts, I am getting error "Retransmission timeout reached on transmission uG3yl-xfLfsJ5aSGwXajJQ.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 17793ms with no response ".

I also checked the logs after turning Sip debug on but I am not able understand why the re-transmission happening. I really need to fix this issue/

Below is the sip debug logs for your reference.

<--- SIP read from UDP:10.200.33.126:5066 --->
INVITE sip:[email protected]:5066;transport=UDP SIP/2.0
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-858f22890add152c-1---d8754z-
Max-Forwards: 70
Contact: <sip:3353@ISP1_Public_IP:5066;transport=UDP>
To: <sip:[email protected]:5066;transport=UDP>
From: "3353"<sip:[email protected]:5066;transport=UDP>;tag=35468a11
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Allow-Events: presence, kpml
Content-Length: 329

v=0
o=Zoiper_user 0 0 IN IP4 ISP1_Public_IP
s=Zoiper_session
c=IN IP4 ISP1_Public_IP
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 10.200.33.126:5066 (NAT)
Sending to 10.200.33.126:5066 (NAT)
Using INVITE request as basis request - M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
Found peer '3353' for '3353' from 10.200.33.126:5066

<--- Reliably Transmitting (no NAT) to 10.200.33.126:5066 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-858f22890add152c-1---d8754z-;received=10.200.33.126
From: "3353"<sip:[email protected]:5066;transport=UDP>;tag=35468a11
To: <sip:[email protected]:5066;transport=UDP>;tag=as46c5438d
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 1 INVITE
Server: FPBX-13.0.195.28(13.28.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33a37f46"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.' in 16448 ms (Method: INVITE)

<--- SIP read from UDP:10.200.33.126:5066 --->
ACK sip:[email protected]:5066;transport=UDP SIP/2.0
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-858f22890add152c-1---d8754z-
Max-Forwards: 70
To: <sip:[email protected]:5066;transport=UDP>;tag=as46c5438d
From: "3353"<sip:[email protected]:5066;transport=UDP>;tag=35468a11
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.200.33.126:5066 --->
INVITE sip:[email protected]:5066;transport=UDP SIP/2.0
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-315ac7e2c554f704-1---d8754z-
Max-Forwards: 70
Contact: <sip:3353@ISP1_Public_IP:5066;transport=UDP>
To: <sip:[email protected]:5066;transport=UDP>
From: "3353"<sip:[email protected]:5066;transport=UDP>;tag=35468a11
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.43 r24984
Authorization: Digest username="3353",realm="asterisk",nonce="33a37f46",uri="sip:[email protected]:5066;transport=UDP",response="f6f9835da16e3a02679e2681fff21fb5",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 329

v=0
o=Zoiper_user 0 0 IN IP4 ISP1_Public_IP
s=Zoiper_session
c=IN IP4 ISP1_Public_IP
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 10.200.33.126:5066 (no NAT)
Using INVITE request as basis request - M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
Found peer '3353' for '3353' from 10.200.33.126:5066
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|ulaw|alaw|gsm|g723), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port ISP1_Public_IP:8000
Looking for 3331 in from-internal (domain 10.1.0.20)
sip_route_dump: route/path hop: <sip:3353@ISP1_Public_IP:5066;transport=UDP>

<--- Transmitting (no NAT) to 10.200.33.126:5066 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-315ac7e2c554f704-1---d8754z-;received=10.200.33.126
From: "3353"<sip:[email protected]:5066;transport=UDP>;tag=35468a11
To: <sip:[email protected]:5066;transport=UDP>
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 2 INVITE
Server: FPBX-13.0.195.28(13.28.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:3331@PBX_Public_IP:5066>
Content-Length: 0


<------------>
Audio is at 20870
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.200.33.201:43430:
INVITE sip:3331@ISP2_Public_IP:43430;rinstance=ac84e9583c6d15d2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP PBX_Public_IP:5066;branch=z9hG4bK6583b52e
Max-Forwards: 70
From: "IT Support" <sip:3353@PBX_Public_IP:5066>;tag=as59a14a7d
To: <sip:3331@ISP2_Public_IP:43430;rinstance=ac84e9583c6d15d2;transport=UDP>
Contact: <sip:3353@PBX_Public_IP:5066>
Call-ID: 19df596d734e5793236b506a59a0d519@PBX_Public_IP:5066
CSeq: 102 INVITE
User-Agent: FPBX-13.0.195.28(13.28.0)
Date: Fri, 20 Sep 2019 15:22:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "IT Support" <sip:3353@PBX_Public_IP>
Content-Type: application/sdp
Content-Length: 392

v=0
o=root 2128058025 2128058025 IN IP4 PBX_Public_IP
s=Asterisk PBX 13.28.0
c=IN IP4 PBX_Public_IP
t=0 0
m=audio 20870 RTP/AVP 0 18 8 3 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- Transmitting (no NAT) to 10.200.33.126:5066 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ISP1_Public_IP:5066;branch=z9hG4bK-d8754z-315ac7e2c554f704-1---d8754z-;received=10.200.33.126
From: "3353"<sip:[email protected]:5066;transport=UDP>;tag=35468a11
To: <sip:[email protected]:5066;transport=UDP>;tag=as4b9c2682
Call-ID: M2Q5MDYwMGZjNjYzOThhMzNlMTMxZGI3ZTY0N2RlMWE.
CSeq: 2 INVITE
Server: FPBX-13.0.195.28(13.28.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:3331@PBX_Public_IP:5066>
P-Asserted-Identity: "Kapil Dev Sharma" <sip:[email protected]>
Content-Length: 0

 

[atsak]

No audio is a firewall issue. Disable SIP ALG and make sure ports 10 000 - 20 000 UDP are forwarding to your PBX.
Also make sure the localnetworks and external IP settings are correct in the SIP settings.

 

[vchohan]

No audio is a firewall issue. Disable SIP ALG and make sure ports 10 000 - 20 000 UDP are forwarding to your PBX.
Also make sure the localnetworks and external IP settings are correct in the SIP settings.

Thank you for your response.
I have checked the firewall and can see SIP ALG is already disabled. We are using RTP pkt range 20 001 to 21 000 and same are allowed on the firewall. Could you please confirm if we can use such a range (20 001 to 21 000) in incredible PBX?

local networks and external IP setting are also correct. Is there any way to check why the re-transmission is happening?

 

[atsak]

Thank you for your response.
I have checked the firewall and can see SIP ALG is already disabled. We are using RTP pkt range 20 001 to 21 000 and same are allowed on the firewall. Could you please confirm if we can use such a range (20 001 to 21 000) in incredible PBX?

local networks and external IP setting are also correct. Is there any way to check why the re-transmission is happening?

I can't think of any reason why you couldn't use 20,000 to 21000, though I'm not sure why you're inclined to change them from default. Did you adjust the iptables firewall settings (etc/sysconfig/iptables usually) to allow those ports and the SIP settings in the GUI?

but this is a firewall issue one way or another.

 

[vchohan]

I can't think of any reason why you couldn't use 20,000 to 21000, though I'm not sure why you're inclined to change them from default. Did you adjust the iptables firewall settings (etc/sysconfig/iptables usually) to allow those ports and the SIP settings in the GUI?

but this is a firewall issue one way or another.

We are using another PBX server which is using UDP port range 10000 -20000.

Thank you so much there were some items missing in the IPtables.