Voice Drop with PJSIP / iridium calls

Dear All,

For convenience, I will name the user calling from Iridium “Speaker A” and user receiving the call “Speaker B”.

Speaker A ó Smart Phone ó SIP application ó WiFi ó Asterix (in gateway) ó Alsa sound system (In gateway) ó I2S (in gateway) ó 9770 Module ó Iridium certified Antenna ó Iridium satellite ó Telecom Operator network ó Phone ó Speaker B

Scenario 1: Fail

  • Speaker A is calling Speaker B: OK
  • Speaker A is talking: good sound
  • Speaker B is answering while Speaker A is silent for more than 5 second: Good Sound
  • Speaker A is replying: Sound is getting worst afterward
  • While Speaker B is always good before and after

Scenario 2: OK

  • Speaker A is calling Speaker B: OK
  • Speaker A is talking: good sound
  • Speaker B is answering while Speaker A is silent for less than 5 second: Good Sound
  • Speaker A is replying: good sound

Scenario 2: OK

  • Speaker A is calling Speaker B: OK
  • Speaker A is talking: good sound
  • Speaker B is answering while Speaker A is in mute (less or more than 5 seconds): Good Sound
  • Speaker A is replying: good sound

Scenario 3: OK

  • Speaker A is calling Speaker B: OK
  • Speaker A is talking for 10 min (no reply from Speaker B): good sound

Scenario 4: OK

  • Speaker A is calling Speaker B: OK
  • Speaker B is talking for 10 min (no reply from Speaker A): good sound

Scenario 5: OK

  • Speaker A is calling Speaker B: OK
  • Speaker A and Speaker B are talking at the same time (no silence from Speaker A): good sound

Scenario 6: OK

  • Speaker A is calling Speaker B: OK

  • Speaker A is talking: good sound

  • Speaker A is silent for more than 5 seconds

  • Speaker A is talking again: good sound

  • We have to combine “5 seconds silence” from Speaker A and talk from Speaker B to trig the problem.

The same SIP application on the smart phone and Asterix in the gateway are also handling a LTE Voice call with the same path as Iridium without facing any problem.

We conclude that our SIP application and Asterix are OK. The only difference is that our LTE module interface is using USB when Iridium 9770 module is using I2S for both ways communication.

Speaker A ó Smart Phone ó SIP application ó WiFi ó Asterix (in gateway) ó Alsa sound system (In gateway) ó USB (in gateway) ó LTE Module ó Telecom Operator network ó Phone ó Speaker B

We also concluded that the hardware and communication with satellite is good as speaker A and Speaker B sound can be good during more than 10 minutes without problem.

It sounds that there is a feature detecting a “silence from speaker A” of more than 5 seconds while “Speaker B is speaking” which impact the quality of the voice of speaker A afterward.

Please help me identify where the issue is.

Here is my pjsip.conf
[global]
type=global
user_agent=PBX

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[7001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=7001
aors=7001
direct_media=no
force_rport=yes
rewrite_contact=yes

[7001]
type=auth
auth_type=userpass
password=7001
username=7001

[7001]
type=aor
remove_existing=yes
max_contacts=1

[7002]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=7002
aors=7002
direct_media=no
force_rport=yes
rewrite_contact=yes

[7002]
type=auth
auth_type=userpass
password=1234qwer
username=7002

[7002]
type=aor
remove_existing=yes
max_contacts=1
authenticate_qualify=yes
qualify_frequency=60

[4G]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=4G
aors=4G
direct_media=no
force_rport=yes
rewrite_contact=yes

[4G]
type=auth
auth_type=userpass
password=1234qwer
username=4G

[4G]
type=aor
remove_existing=yes
max_contacts=1

[iridium]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=iridium
aors=iridium
direct_media=no
force_rport=yes
rewrite_contact=yes

[iridium]
type=auth
auth_type=userpass
password=1234qwer
username=iridium

[iridium]
type=aor
remove_existing=yes
max_contacts=1

; Jitter Buffer
rtp_symmetric=yes
rtp_timeout=60

; 4G Jitter Buffer
[4G]
jitterbuffer=yes
jitterbuffer_size=50 ;
jitterbuffer_resync_threshold=1000 ;

[iridium]
jitterbuffer=yes
jitterbuffer_size=10 ;
jitterbuffer_resync_threshold=10 ;

question who is having problem hearing A or B
can you descripe how the degraded sound is like

  • robot
  • choppy
  • migimouse

also what is the bandwith / latency for Iridium ?

if posible can you try capture / enable rtcp packages
https://wiki.wireshark.org/RTCP

Google says 395ms ± 100 milliseconds. I think that is one way (one up- and one down-link).

Is this Iridium voice or Iridium broadband? The former is 2.4kbps, which is never going to sound good.

Dear All,
the issue is on iridium FW.
Iridium provided the latest firmware to us and the issue has been resolved.
The other thing I can share here is that the setting of the jitter buffer can help improve the call quality.

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