Call mutes after 5 seconds on gxp2170 but not on wp820


#1

I have a gxp 2170 that can take or make a call but loses sound after 5 seconds. The line stays active and the session timer is still going but no sound can go through either side. If I set this exact same scenario up on my wp820 it works just fine. From what I can tell the phones are set up mostly the same, vanilla from the box for the most part.
Our pbx is an Toshiba ipedge server. These lines are both registered as sip. I really don’t see how it could be on the phone server as the wp820 works just fine. I almost tired this with a gxp 1625 and it will mute the call after 5 seconds too. It only happens when this is from a different phone type, the grandstream phones all can talk to each other with no interruptions at all, I only need this functionality to work while we do the changeover but still it is needed. Please help.


#2

stun?
NAT?
Kepp alive?
We need more details.


#3

Stun is just set to default in the phones whatever they came with. NAT isn’t used. Keep alive is set to 20 Sorry I am far from an expert on these things.

edit Stun blank, nat blank, keep alive 20


#4

the stun I mean disabled? I hope so.
Trying to change the codec? Just put PCMU/PCMA on both the server side and the phones.
Have you also contacted Toshiba Support? (I think Mitel)


#5

Yes stun is disabled.
I tried changing the codecs around it doesn’t seem to matter.
If it was something to do with the server, why does the wp820 work just fine?


#6

Was the 820 and 2170 tested using the Toshiba and with the same extension number for both devices? In the phone, Account, Network, NAT should be No if on the same LAN.

The server can be a factor as well as it has settings that accommodate the phones locations and if the phones are allowed to direct audio or if the phone system itself will handle the audio.


#7

Just tested it. I changed the sip ext from the 2170 to the 820 and it worked fine. Could be something to do with direct audio, it is odd that you sound perfectly fine for 5 seconds and it cuts it. I am thinking its some kind of handoff of the call that


#8

Only a network capture at the device or PBX would tell what is happening, but as I am not familiar with the Toshiba extension settings, you can trying comparing and see if you can figure it out.


#9

This is a capture at the device of the udp stream

INVITE sip:248@128.128.120.3:5060 SIP/2.0

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d;received=128.128.0.40

Max-Forwards: 70

Record-Route: <sip:128.128.0.40>

To: 248 <sip:248@128.128.0.40>

From: "IT SERVER ROOM" <sip:331@128.128.0.40;user=phone>;tag=fbf5d52b70f5a69d

Call-ID: 8000006615d52b70f5a6b9@128.128.0.40

CSeq: 25 INVITE

Contact: 331 <sip:331@128.128.0.40;user=phone>

Content-Type: application/sdp

Content-Length: 216

Supported: replaces

v=0

o=331 1226511370 1226511370 IN IP4 128.128.0.40

s=-

c=IN IP4 128.128.0.40

t=0 0

m=audio 18672 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d;received=128.128.0.40

From: "IT SERVER ROOM" <sip:331@128.128.0.40;user=phone>;tag=fbf5d52b70f5a69d

To: 248 <sip:248@128.128.0.40>

Call-ID: 8000006615d52b70f5a6b9@128.128.0.40

CSeq: 25 INVITE

Supported: replaces, path, timer

User-Agent: Grandstream GXP2170 1.0.9.135

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE

Content-Length: 0

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d;received=128.128.0.40

Record-Route: <sip:128.128.0.40>

From: "IT SERVER ROOM" <sip:331@128.128.0.40;user=phone>;tag=fbf5d52b70f5a69d

To: 248 <sip:248@128.128.0.40>;tag=31948431

Call-ID: 8000006615d52b70f5a6b9@128.128.0.40

CSeq: 25 INVITE

Contact: <sip:248@128.128.120.3:5060>

Supported: replaces, path, timer

User-Agent: Grandstream GXP2170 1.0.9.135

Allow-Events: talk, hold

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE

Content-Length: 0

SIP/2.0 200 OK

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK5b4e3c17f70af9664bb13404e8f6852d;received=128.128.0.40

Record-Route: <sip:128.128.0.40>

From: "IT SERVER ROOM" <sip:331@128.128.0.40;user=phone>;tag=fbf5d52b70f5a69d

To: 248 <sip:248@128.128.0.40>;tag=31948431

Call-ID: 8000006615d52b70f5a6b9@128.128.0.40

CSeq: 25 INVITE

Contact: <sip:248@128.128.120.3:5060>

Supported: replaces, path, timer

User-Agent: Grandstream GXP2170 1.0.9.135

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE

Content-Type: application/sdp

Content-Length: 212

v=0

o=248 8000 8000 IN IP4 128.128.120.3

s=SIP Call

c=IN IP4 128.128.120.3

t=0 0

m=audio 5016 RTP/AVP 0 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

ACK sip:248@128.128.120.3:5060 SIP/2.0

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bKef6006510c0e1b7557b663014efc20a0

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bKef6006510c0e1b7557b663014efc20a0;received=128.128.0.40

Max-Forwards: 70

Record-Route: <sip:128.128.0.40>

From: "IT SERVER ROOM" <sip:331@128.128.0.40;user=phone>;tag=fbf5d52b70f5a69d

To: 248 <sip:248@128.128.0.40>;tag=31948431

Call-ID: 8000006615d52b70f5a6b9@128.128.0.40

CSeq: 25 ACK

Content-Length: 0

OPTIONS sip:248@128.128.120.3:5060 SIP/2.0

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK6d614025851899e1666fb100ed6cfb67

Max-Forwards: 70

Record-Route: <sip:128.128.0.40>

To: 248 <sip:248@128.128.0.40>

From: 248 <sip:248@128.128.0.40>;tag=7de5d52b718b1c28

Call-ID: 8000006665d52b718b1c3b@128.128.0.40

CSeq: 26 OPTIONS

Content-Length: 0

SIP/2.0 200 OK

Via: SIP/2.0/UDP 128.128.0.40:5060;branch=z9hG4bK6d614025851899e1666fb100ed6cfb67

From: 248 <sip:248@128.128.0.40>;tag=7de5d52b718b1c28

To: 248 <sip:248@128.128.0.40>;tag=888429181

Call-ID: 8000006665d52b718b1c3b@128.128.0.40

CSeq: 26 OPTIONS

Supported: replaces, path, timer

User-Agent: Grandstream GXP2170 1.0.9.135

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE

Content-Length: 0


#10

please post the pcap file itself. It helps to be able to see the flow rather than trying to read a serial log.


#11

packetstest.zip (37.8 KB)


#12

WP820 is a VoIP terminal works differently from the GXP2170, in addition to being in wifi …


#13

the pcap completely lacks the RTP audio stream
I also see only 2 ip, missing the third, the server voip what ip has?


#14

test2wrtp.zip (82.6 KB)
Sorry here is rtp. VOIP server is 128.128.0.40


#15

once such a thing happened to me, it was the server that tried to peer the audio of the 2 terminals, try to disable it

  • on UCM is “Can Direct Media”.
  • on 3cx “PBX Delivers Audio”
    I have no idea about Toshiba.

#16

since with WP820 audio is regular, do you send an inherent pcap?


#17

Need to know a little more about the scenario.

What device was used to place the call?
We apparently only see the leg between the server and the 2160, and there is audio. What I do not see if what device is on the other side of the PBX that placed the call. I only know that ext 331 called 248, but the device that actually made the call is not exposed unless possibly it was a softphone on the PBX itself, but no UA showed itself.


#18

in fact, the third interpreter is missing.


#19

Thanks for the help guys I do appreciate it. You are making this situation more clear. I agree it might be upwards more actually in the pbx system we have. So to give you a top down view of the project, we have an old Toshiba cix670 system with a whole bunch of slt phones tied to it along with several other devices. We also have a new Toshiba ipedge server, which is what we are attempting to tie these new gxp2170’s into. The problem seems to be coming from when we try to make a call from or to the old cix system. The only odd part is the wp820’s don’t have the issue while the 2170’s do. The cix system is 128.128.9.41


#20

Tested the wave app and it works fine for each type of call just as the 820 does. Makes me again wonder what the 2170 is doing different. I am going through flipping on and off every setting in the device.