sundance-communications.com/forum Forums TDM Business Telephone Systems Samsung Telephones & Systems SIP delay on start RTP

Hi all,
I realized there no audio in the first 4 seconds on SIP, then the conversation goes on correctly. Any idea how to solve this issue?

Regards.

More information required.

System type and version
Handset type
Are there multiple MGI/OAS cards?
Ports forwarded correctly in the router?

This happens in almost all systems with SIP settings, but i'm referring to an OS7200 MCP v4.46a, OAS v2.01.

You need to get a trace from the system side to see if the RTP stream is getting to the OAS card, and what ports it is trying to reach.

If you can get your carrier to do a trace from their end at the same time and you can compare to see if the router is the issue or the system.

The problem is not with a SIP carrier. Is a connection to an IVR through SIP exten.

interface: any
filter: (ip) and ( port 5060 )
#
U 2011/06/21 13:33:45.254195 192.168.7.100:5060 -> 192.168.7.25:5060
INVITE sip:[email protected] SIP/2.0
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.7.100:5060;rport;branch=z9hG4bK-3dc8af4a-57ecbb44-17f5461
Max-Forwards: 70
Supported: 100rel,replaces
Reason: SIP; cause=1; text="0916326542"
Contact: <sip:[email protected]:5060>
Content-Type: application/SDP
Content-Length: 229

v=0
o=SAMSUNG_SIP_GATEWAY 1475132332 0 IN IP4 192.168.7.100
s=SIP_CALL
c=IN IP4 192.168.7.101
t=0 0
m=audio 30000 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

#
U 2011/06/21 13:33:45.257186 192.168.7.25:5060 -> 192.168.7.100:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.100:5060;branch=z9hG4bK-3dc8af4a-57ecbb44-17f5461;received=192.168.7.100;rport=5060
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0


#
U 2011/06/21 13:33:45.257244 192.168.7.25:5060 -> 192.168.7.100:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.100:5060;branch=z9hG4bK-3dc8af4a-57ecbb44-17f5461;received=192.168.7.100;rport=5060
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>;tag=as08b90a39
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 252344691 252344691 IN IP4 192.168.7.25
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.7.25
t=0 0
m=audio 15136 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

#
U 2011/06/21 13:33:46.255826 192.168.7.25:5060 -> 192.168.7.100:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.100:5060;branch=z9hG4bK-3dc8af4a-57ecbb44-17f5461;received=192.168.7.100;rport=5060
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>;tag=as08b90a39
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 252344691 252344691 IN IP4 192.168.7.25
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.7.25
t=0 0
m=audio 15136 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

#
U 2011/06/21 13:33:47.255704 192.168.7.25:5060 -> 192.168.7.100:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.100:5060;branch=z9hG4bK-3dc8af4a-57ecbb44-17f5461;received=192.168.7.100;rport=5060
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>;tag=as08b90a39
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 252344691 252344691 IN IP4 192.168.7.25
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.7.25
t=0 0
m=audio 15136 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

#
U 2011/06/21 13:33:49.255532 192.168.7.25:5060 -> 192.168.7.100:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.100:5060;branch=z9hG4bK-3dc8af4a-57ecbb44-17f5461;received=192.168.7.100;rport=5060
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>;tag=as08b90a39
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 252344691 252344691 IN IP4 192.168.7.25
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.7.25
t=0 0
m=audio 15136 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

#
U 2011/06/21 13:33:49.559793 192.168.7.100:5060 -> 192.168.7.25:5060
ACK sip:[email protected] SIP/2.0
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>;tag=as08b90a39
Call-ID: [email protected]
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.7.100:5060;rport;branch=z9hG4bK-3dc8af4e-57eccc8c-52f2ab86
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
Content-Length: 0


#
U 2011/06/21 13:33:52.202120 192.168.7.100:5060 -> 192.168.7.25:5060
ACK sip:[email protected] SIP/2.0
From: <sip:[email protected]>;tag=c0a80764-13c4-3dc8af4a-57ecbb42-1e41fbc8
To: <sip:[email protected]>;tag=as08b90a39
Call-ID: [email protected]
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.7.100:5060;rport;branch=z9hG4bK-3dc8af4e-57eccc8c-52f2ab86
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
Content-Length: 0


exit
8 received, 0 dropped

both systems are in the same network.

If i read this correctly 192.168.7.100 is the MCP
192.168.7.101 is the OAS card and 192.168.7.25 is the asterisk box.

First packet the OS is making a call to the asterisk extension "produccion", this also tells the asterisk box to use PCMA (G.711 a-law) codec, and to send audio to 192.168.7.101 on port 30000.

Then we have a progress message from asterisk to os7200

Then we have a ok message from the asterisk box which also tells the OS7200 to send audio to 192.168.7.25 on port 15136 (PCMA Codec)

Etc

You have only captured the signalling which looks ok. Can you get another one with the audio packets as well?