F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

VoipServer Card ( Arm AT91SAM9G45 with Linux )

1
14
ME:- C_101_036_01_01.jed (8 Channel)
C_101_035_01_01.jed (16 Channel)
C_101_034_01_01.jed (24 Channel)
C_101_024_01_01.jed (32 Channel)
GE:- C_100_033_02_01.jed (8 Channel)
C_100_032_02_01.jed(16 Channel)
C_100_031_02_01.jed (24 Channel)
C_100_020_02_01.jed (32 Channel)
PE:- C_138_024_01_01.jed (8 Channel)
C_138_013_01_01.jed(16 Channel)
Compatible with Eternity V10R11 & Voip Server V1R10
& onwards,
ME: C_101_024_01_02.jed (32 Channels)
C_101_034_01_02.jed (24Channels)
C_101_035_01_02.jed (16 Channels)
C_101_036_01_02.jed (8 Channels)
GE: C_100_020_02_02.jed (32 Channels)
C_100_031_02_02.jed (24Channels)
C_100_032_02_02.jed (16 Channels)
C_100_033_02_02.jed (8 Channels)
PE:- C_138_024_01_02.jed (8 Channel)
C_138_013_01_02.jed(16 Channel)

Microcontrollers Version/Revision (if applicable)

ME:-P-101-024-01-02
GE:- P-100-020-02-02 & onwards.
PE:- P-138-013-01-02
N.A
For Production:
flashloader.bin
image.bin
uImage
VoipServer_V1R14.jffs2
VoipServer_V1R14.bin

F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

VoipServer Card ( Arm AT91SAM9G45 with Linux )

For Field Upgrade:
490008ae3.660.17
490k160.660.17; 246
490mt.660_3.17; 956
awk
ax88796b.ko
busybox
CA.sh
CreateCertificate.sh
ez-ipupdate
GpioDrv.ko
inittab
ipcQueue
md5sum
media
openssl
opensslCa.cnf
openssl.cnf
passwd
rcS
sysLog
uImage
voIPServer
VoipServerApp
voipserver.sh
WatchDogClient
(File sixe in bytes)
For Production:
flashloader.bin
image.bin
uImage
VoipServer_V1R14.jffs2
VoipServer_V1R14.bin

7540
524288
1401228
6422528
33554432

For Field Upgrade:

490008ae3.660.17
490k160.660.17

368972
246

F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

VoipServer Card ( Arm AT91SAM9G45 with Linux )

490mt.660_3.17
956
awk
453296
ax88796b.ko
21635
busybox
453296
CA.sh
4033
CreateCertificate.sh
772
ez-ipupdate
129367
GpioDrv.ko
8369
inittab
235
ipcQueue
6759
md5sum
453296
media
337572
openssl
1668260
opensslCa.cnf
2829
openssl.cnf
2831
passwd
453296
rcS
374
sysLog
13251
uImage
1401228
voIPServer
1023
VoipServerApp
4521312
voipserver.sh
734
WatchDogClient
9712
9 - Sep-2013

F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

For Production:
flashloader.bin
(5f9c6bb1209e914410da949ca54bbdfe)
image.bin
(ff5a73cff231e7b8f9bcf15140cf5452)
uImage
(1eb1a2624f160aeb70624f94a53277cd)
VoipServer_V1R14.jffs2
(7f10b24c380cce2af5b1ba9f7b391ee2)
VoipServer_V1R14.bin
(27e62173414098c87237f3afc5725897)
For Field Upgrade:
490008ae3.660.17
(01de44cff5d460a06096bd26e4d0fcca)
490k160.660.17
(2f695f5c68f1b41187f2d11000fb143f)
490mt.660_3.17
(ab9442be5395e7cfc4f7ed80d48424ab)
Awk
(5f7cad1e78b5c0aeb1f822626bd43c5b)
ax88796b.ko
(62c71b0e5846e5ae9b3355aa886921f9)
Busybox
(8d79468a8e3c15482d76cb55a66f2d77)
CA.sh
(98de73a856a195b0ebbea5d0d87212bd)
CreateCertificate.sh
(c8fed5ff3b2419ca9879723bc38199c0)
ez-ipupdate
( a58cf8fb7bc4ca78308aa0e6edc34f6a)
GpioDrv.ko
( 7346b8ad4d963998ae4d91640a1a5624)
Inittab
(b7c8b0b4abad5bc937b6efe76e7d1b67)
ipcQueue
(718a1c4a27f19060bf9a7f57bff3733f)
md5sum
(8d79468a8e3c15482d76cb55a66f2d77)
media
(27ec301f00e8843482648e80f059abf4)
Openssl
(6b2841b7f4279f3a75e450023eb315c3)
opensslCa.cnf

F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

(ef4c609dd56094004657bd47b6eb3038)
openssl.cnf
(5fda4df274f837654c8711eb3131753c)
Passwd
( 8d79468a8e3c15482d76cb55a66f2d77)
rcS
(c342dae5d02e156fa3d207e720184d7f)
sysLog
(2bbefa0fe8b8e2025208412ac51d4534)
uImage
(1eb1a2624f160aeb70624f94a53277cd)
voIPServer
(90ad8f6f12a465d5795e4444ca63d0d7)
VoipServerApp
(2e653c186cccba351ff4ab839db1fc7f)
voipserver.sh
(1cf5364bf3c3860e40ee4931ccfc5a37)
WatchDogClient
(1b84fbdc3086622f2714300d728af2d9)
a)Will this firmware up gradation affect the system
configuration?
b)Will the system automatically set to default
configuration?
c)Give details of configuration changes after upgradation

NA

NA
NA
Version / revision of SAD document to which this release
complies to
(Mention exceptions, if any)

Rev 0.159 SIP User Registration from different subnet

svn://192.168.100.5/Briefcase/SAD_Briefcase/Knowled
ge Bank/VoIP_Cases.
1) Use dynamic port for STUN request
Check Release Note of Master V11R1
NA
Yes

F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

New Features Implementation:

Following feature will work with V10R10 & onwards with VoIP Server V1R14.
1.Silence suppression flag is only applicable for G.729 codec.
2.Problem: In Portugal, its found that sometimes STUN request from Eternity gets blocked by
the local router. Reason could be Eternity uses same port (either 1024 or 1025) for periodic
STUN query and due to overflow of packets, router blocks the further packets from Eternity. At
the same time, its observed that STUN requests from softphone gets served by the router as
softphone sends periodic STUN query from dynamic port.
Solution: Use dynamic port for periodic STUN query.
3.VoIP application was restarted while changing ST bus at run time. Now it will not restart on
changing ST bus.
Following feature will work with V11R1 & onwards with VoIP Server V1R14.
1.Video call support.
2.VoIP Server card auto Hardware test is now supported.
3.SPARSH MS for Iphone Support is added.
4.All call reject responses are now passed to master application for SIP-ISDN & SIP-GSM cause
mapping.
5.Maximum 10 White list IP entry is now supported.
Problems Solved:
1.Source Port IP Address is programmed as IP Address Fetched using STUN/Router's Public IP
Address in SIP Extension General Parameter page. If INVITE is received from different subnet
with WAN IP in ReqURI field of SIP message, it was rejected by system. Now it is solved by
implementing Rev 0.159.
2.PRACK was received in UDP though INVITE was received in TCP from VoIP client.
Reason: Transport was not set in Contact header of Provisional responses send by VoIP Server.
That was creating problem in PRACK with few clients when used with TCP.
3.When VoIP Server was sending BYE due to timeout for not getting ACK against 200Ok of
INVITE, LAN IP was set in IP header even though request was sent from WAN port.
4.When ACK against 200OK of initial INVITE was not received, few resources in SIP Stack did
not free. System was unable to establish calls after sometime due to such errors.
Symptoms: rv = -2 error can be found in debug during call OR 486/500 response against
INVITE but no event in application log.
5.After few Registration refreshes, system stops sending REGISTER message for TRUNK when
domain is configured. Once problem starts, its occurrence is random.
Problem:Blank IP was received for Domain while sending request (Registration). DNS was
resolved but SIP Stack was not giving IP to application.
6.When Gateway is changed, previous Gateway was not getting removed from VoIP status page.
7.System was hanged when iLBC codec is found in codec list and fmtp parameters having more
than 8 characters (e.g. a=fmtp:96 usedtx=1). This issue was observed on random basis.

F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

Increased buffer size from 8 to 50 to solve this issue.

8.If user is registered with TCP on WAN port and we try to send IM to that user on Master's
request than packet was sent from LAN IP and with UDP transport.
Problem: Transport was not set in ReqURI.
9.If publish is not received, IM was sent on first location if user is registered. But if first location
is blank and user is registered on 2nd location, we could not forward IM message to actual
registered location.
Problem: We were using first location without checking active location.
10.Packet loss was observed in PCAP captured on system when multiple end points are in speech
(more than 2 end points). It is fine tune now and PCAP is working fine till 10 audio end points
are in speech.
11.BYE was sent with Authentication header. Remote party send 401 with new nonce value.
Server card Send BYE with two Authentication header in which first Authentication header was
with old nonce followed by header with new nonce. Hence request was not accepted by remote
end and it sends 401 again.
12.Video call related issue solved which was observed in previous V1R13 release is solved.
Problem: Unable to open video RTP port which cause no video stream transfer between two
video terminal in video call.
Limitations:
1.Audio Codes DSP module does not support pTime (Packet Time) above 40ms. It will use
pTime as 20ms for that.
2.VoIP Server does not support 200 OK with SDP against received OPTION (SIP Server stack
limitation).
3.T38 RTP fax is not working between audio codes version 640 (VoIP Server Version V1R7 for
Eternity GE/ME/PE) to 660 (VoIP Server Version V1R8 & V1R10 for Eternity GE/PE/ME) &
vice versa, but working fine between audio codes version 660 (VoIP Server Version V1R8 &
V1R10 for Eternity GE/ME/PE).

Notes for Video Call feature:

1.Our system can/will not modify any parameter in Video stream (alt etc...).
2.Disable ICE feature in phone for video calling. ICE is having facility to convey ip address
within m line (Apart from connection information). This will cause video flow between two
endpoints directly. It can create lip sync issue.
3.SRTP will disable internally when video media stream is received.
4.SSRC, Timestamp & Sequence number of RTP will change when switch in from Video call to
Hold and Hold to normal Video call.
5.We are not supporting VFU in INFO (XML format). Side effect of it is still not known to us.
6.During hold in Video call, media attribute in Video stream will be send as it is (which was
received) as we are not touching Video streams.

F-R&D-SWD-10

Revised on: 15-10-2010

Revision No: 00
SWD MANU/SUPP

Caution:
1.Do not enable VoPP debug in VoIP Debug page unless suggested by Matrix R&D engineer.
This shall deteriorate the performance of Card. It is for internal use only.
Notes:
1.This Version Revision will work with only Eternity V10R10 & onwards.
2.VoIP Server Application restart as change in Static Routing Table.
3.For Eternity ME use ETME_SLAVE_CM_V1R2.TSK Eternity ME slave card Comm
Manager.

Dhaval Patel

9- Sep -2013