Pjlib on Symbian
Hi,
I created an application based on pjlib on Symbian OS and have some problems
with the sound.
The SIP session is OK but during the media session the sound is interrupted
at the caller side and the destination has an echo (during of all
conversation) and the sound is interrupted also.
I did the tests with Nokia E71 on 3G network.
Do we need a 3G network?
What are the minimum requirements (Phone hardware: processor, memory and
others) to have a good quality of sound?
Thanks,
George.
2009/3/17 George Evi george.evi@ctcinc.ca
Hi,
I created an application based on pjlib on Symbian OS and have some
problems with the sound.
The SIP session is OK but during the media session the sound is interrupted
at the caller side and the destination has an echo (during of all
conversation) and the sound is interrupted also.
By "interupted", did you mean like stuttering? Does it happen often?
If it is stutter, it could be caused by network jitter, or some activity in
the application (we found that even simple activity such as printing log
message to console screen could delay the audio).
Regarding the echo, I realize that the echo suppressor in pjmedia is still
work in progress, so "a bit" of echo is quite expected.
I did the tests with Nokia E71 on 3G network.
Do we need a 3G network?
I would say Wi-Fi would work better.
What are the minimum requirements (Phone hardware: processor, memory and
others) to have a good quality of sound?
Minimum requirement is an S60 3rd ed device. For best quality, use
APS-Direct ([1], to be included in release 1.1). It uses native/handset's
codec and echo canceller, and in my personal test there is zero echo with
this. Though the drawback is it needs Symbian signing.
cheers
Benny
Thanks,
George.
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected Wi-Fi
and I expected to see some improvements but voice stilled disrupted.
I'm using iLBC as codec (1st priority) and UDP transport.
Do you have any suggestions?
Thank you,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Thursday, March 19, 2009 11:27 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/17 George Evi george.evi@ctcinc.ca
Hi,
I created an application based on pjlib on Symbian OS and have some problems
with the sound.
The SIP session is OK but during the media session the sound is interrupted
at the caller side and the destination has an echo (during of all
conversation) and the sound is interrupted also.
By "interupted", did you mean like stuttering? Does it happen often?
If it is stutter, it could be caused by network jitter, or some activity in
the application (we found that even simple activity such as printing log
message to console screen could delay the audio).
Regarding the echo, I realize that the echo suppressor in pjmedia is still
work in progress, so "a bit" of echo is quite expected.
I did the tests with Nokia E71 on 3G network.
Do we need a 3G network?
I would say Wi-Fi would work better.
What are the minimum requirements (Phone hardware: processor, memory and
others) to have a good quality of sound?
Minimum requirement is an S60 3rd ed device. For best quality, use
APS-Direct ([1], to be included in release 1.1). It uses native/handset's
codec and echo canceller, and in my personal test there is zero echo with
this. Though the drawback is it needs Symbian signing.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
Thanks,
George.
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Did you tried different codes?
Like GSM, G729, AMR, G711?
Please consider that the 3G connection and the 'packetizing algorithm'
has to be carefully design on Symbian in order not to incurr in overflow
and underflow.
Did you checked if there's some buffer overflowing or underflowing?
Fabio
George Evi wrote:
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and
on the callee side the voice is metalique (like a robot speech). The
latest tests I made were done with Nokia E61 (S60 3^rd edition -mr)
connected Wi-Fi and I expected to see some improvements but voice
stilled disrupted.
I’m using iLBC as codec (1^st priority) and UDP transport.
Do you have any suggestions?
Thank you,
George.
From: pjsip-bounces@lists.pjsip.org
[mailto:pjsip-bounces@lists.pjsip.org] *On Behalf Of *Benny Prijono
Sent: Thursday, March 19, 2009 11:27 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/17 George Evi <george.evi@ctcinc.ca mailto:george.evi@ctcinc.ca>
Hi,
I created an application based on pjlib on Symbian OS and have
some problems with the sound.
The SIP session is OK but during the media session the sound is
interrupted at the caller side and the destination has an echo
(during of all conversation) and the sound is interrupted also.
By "interupted", did you mean like stuttering? Does it happen often?
If it is stutter, it could be caused by network jitter, or some
activity in the application (we found that even simple activity such
as printing log message to console screen could delay the audio).
Regarding the echo, I realize that the echo suppressor in pjmedia is
still work in progress, so "a bit" of echo is quite expected.
I did the tests with Nokia E71 on 3G network.
Do we need a 3G network?
I would say Wi-Fi would work better.
What are the minimum requirements (Phone hardware: processor,
memory and others) to have a good quality of sound?
Minimum requirement is an S60 3rd ed device. For best quality, use
APS-Direct ([1], to be included in release 1.1). It uses
native/handset's codec and echo canceller, and in my personal test
there is zero echo with this. Though the drawback is it needs Symbian
signing.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
Thanks,
George.
_______________________________________________
Visit our blog: http://blog.pjsip.org
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Hi Fabio,
in the project "pjproject-1.0.1" the Symbian project file (mmp) has defined
only 2 codecs: GSM and Speex. I added in my project the "iLBC" codec.
What I understood is during the SDP exchange the SIP phone is sending in its
media description the used codec. I think they are negotiating the codec
usage but I don't know who decide which is the codec to use (the Phone
client or the server).
The last tests I made are with a Nokia E61 connected in Wi-Fi.
I'm new in "pjlib" but as you suggested to look for buffer overflowing or
underflowing, where should I look? In jitter buffer?
Thanks,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Fabio Pietrosanti (naif)
Sent: Thursday, March 19, 2009 1:16 PM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
Did you tried different codes?
Like GSM, G729, AMR, G711?
Please consider that the 3G connection and the 'packetizing algorithm' has
to be carefully design on Symbian in order not to incurr in overflow and
underflow.
Did you checked if there's some buffer overflowing or underflowing?
Fabio
George Evi wrote:
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected Wi-Fi
and I expected to see some improvements but voice stilled disrupted.
I'm using iLBC as codec (1st priority) and UDP transport.
Do you have any suggestions?
Thank you,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Thursday, March 19, 2009 11:27 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/17 George Evi george.evi@ctcinc.ca
Hi,
I created an application based on pjlib on Symbian OS and have some problems
with the sound.
The SIP session is OK but during the media session the sound is interrupted
at the caller side and the destination has an echo (during of all
conversation) and the sound is interrupted also.
By "interupted", did you mean like stuttering? Does it happen often?
If it is stutter, it could be caused by network jitter, or some activity in
the application (we found that even simple activity such as printing log
message to console screen could delay the audio).
Regarding the echo, I realize that the echo suppressor in pjmedia is still
work in progress, so "a bit" of echo is quite expected.
I did the tests with Nokia E71 on 3G network.
Do we need a 3G network?
I would say Wi-Fi would work better.
What are the minimum requirements (Phone hardware: processor, memory and
others) to have a good quality of sound?
Minimum requirement is an S60 3rd ed device. For best quality, use
APS-Direct ([1], to be included in release 1.1). It uses native/handset's
codec and echo canceller, and in my personal test there is zero echo with
this. Though the drawback is it needs Symbian signing.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
Thanks,
George.
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
2009/3/19 George Evi george.evi@ctcinc.ca
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected
Wi-Fi and I expected to see some improvements but voice stilled disrupted.
I’m using iLBC as codec (1st priority) and UDP transport.
Aha, that's probably the reason. iLBC is heavy, I don't think the device has
enough processing power to run it [2]. Try with GSM or Speex.
Alternatively, consider using APS-Direct [1], available in pjsip version 1.1
now downloadable from the website. APS-Direct uses handset's native codec
and it supports iLBC, AMR, G.729, and G.711.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
[2] http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS
Hi Benny,
I update my application with the latest version the "trunk pjproject- 1.1"
and continue to test on Nokia E61.
As you suggested I changed the codec priorities in a way that GSM had
highest priority (in function "pjsua_media_subsys_init" priority value =
PJMEDIA_CODEC_PRIO_NORMAL +4 (132)). The sound was acceptable on the caller
side but on the callee side continue to be stuttered, disrupted and
instable.
Also I tried the others codecs (iLBC, Speex/8000 and Speex/16000) in the
same way but didn't see any improvements.
Do you have any suggestions or ideas?
We don't want to use Nokia APS (Audio proxy Server) for the moment, because
it needs a publisher ID.
Thank you,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Friday, March 20, 2009 2:32 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/19 George Evi george.evi@ctcinc.ca
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected Wi-Fi
and I expected to see some improvements but voice stilled disrupted.
I'm using iLBC as codec (1st priority) and UDP transport.
Aha, that's probably the reason. iLBC is heavy, I don't think the device has
enough processing power to run it [2]. Try with GSM or Speex.
Alternatively, consider using APS-Direct [1], available in pjsip version 1.1
now downloadable from the website. APS-Direct uses handset's native codec
and it supports iLBC, AMR, G.729, and G.711.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
[2] http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS
2009/3/24 George Evi george.evi@ctcinc.ca
Hi Benny,
I update my application with the latest version the “trunk pjproject- 1.1”
and continue to test on Nokia E61.
As you suggested I changed the codec priorities in a way that GSM had
highest priority (in function “pjsua_media_subsys_init*” priority value
= PJMEDIA_CODEC_PRIO_NORMAL +4 (132)*). The sound was acceptable on
the caller side but on the callee side continue to be stuttered, disrupted
and instable.
Hi George,
In that case, I would probably suggest to try to use different peer for the
testing (pjsua running on desktop would be a good candidate :) ). And make
sure the audio doesn't get routed through the server or otherwise this
wouldn't make any difference. You can call directly to the device's IP
address to make sure.
Also I tried the others codecs (iLBC, Speex/8000 and Speex/16000) in the
same way but didn’t see any improvements.
iLBC and Speex/16000 is definitely out of question. Speex/8000 is probably
bang on the processing capability, so use it with care (e.g. only use
release mode), and probably is not good for troubleshooting problems like
this. And of course there is G.711, definitely a good candidate to try.
cheers
Benny
Do you have any suggestions or ideas?
We don’t want to use Nokia APS (Audio proxy Server) for the moment, because
it needs a publisher ID.
Thank you,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:
pjsip-bounces@lists.pjsip.org] *On Behalf Of *Benny Prijono
Sent: Friday, March 20, 2009 2:32 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/19 George Evi george.evi@ctcinc.ca
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected
Wi-Fi and I expected to see some improvements but voice stilled disrupted.
I’m using iLBC as codec (1st priority) and UDP transport.
Aha, that's probably the reason. iLBC is heavy, I don't think the device
has enough processing power to run it [2]. Try with GSM or Speex.
Alternatively, consider using APS-Direct [1], available in pjsip version
1.1 now downloadable from the website. APS-Direct uses handset's native
codec and it supports iLBC, AMR, G.729, and G.711.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
[2] http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi Benny,
I activate the log in my project (PJ_LOG()) and used the function
"log_call_dump()" to dump the statistics at the end of a call.
I got these statistics:
--end msg--State changed from Null to Calling, event=TX_MSGTransaction
tsx0x732e8c state changed to Calling
[CONFIRMED] To: sip:5148403000@sip6.van.netvoice.ca;tag=as5b69601c
Call time: 00h:01m:56s, 1st res in 3542 ms, conn in 5824ms
SRTP status: Not active Crypto-suite: (null)
#0 PCMU @8KHz, sendrecv, peer=64.34.49.82:19122
RX pt=0, stat last update: 00h:00m:00.601s ago
total 5.7Kpkt 924.8KB (1.15MB +IP hdr) @avg=62.3Kbps/77.9Kbps
pkt loss=143 (2.4%), discrd=1 (0.0%), dup=0 (0.0%), reord=1 (0.0%)
(msec) min avg max last dev
loss period: 20.000 178.750 940.000 100.000 63.638
jitter : - 0.001 11.737 579.000 0.750 8.775
TX pt=0, ptime=20ms, stat last update: 00h:00m:03.680s ago
total 5.7Kpkt 906.4KB (1.13MB +IP hdr) @avg 61.1Kbps/76.5Kbps
pkt loss=2 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 40.000 40.000 40.000 40.000 0.000
jitter : 193.000 193.000 193.000 193.000 0.000
RTT msec : 92.000 128.637 332.000 101.000 21.526
Processing incoming message: Response msg 200/BYE/cseq=18255
(rdata0x71e65c)RX 510 bytes Response msg 200/BYE/cseq=18255 (rdata0x71e65c)
from UDP 64.34.49.82:5060:
SIP/2.0 200 OK
I also read the "Understanding Media Flow" document and I have a (beginner)
question.
In the TX section we have a jitter line but in the Media Flow diagram there
is no Jitter Buffer for packet transmission, what represents this line? And
also why in the same section the loss period and jitter buffer values are
the same for all statistics colons?
Thanks,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Wednesday, March 25, 2009 3:56 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/24 George Evi george.evi@ctcinc.ca
Hi Benny,
I update my application with the latest version the "trunk pjproject- 1.1"
and continue to test on Nokia E61.
As you suggested I changed the codec priorities in a way that GSM had
highest priority (in function "pjsua_media_subsys_init" priority value =
PJMEDIA_CODEC_PRIO_NORMAL +4 (132)). The sound was acceptable on the caller
side but on the callee side continue to be stuttered, disrupted and
instable.
Hi George,
In that case, I would probably suggest to try to use different peer for the
testing (pjsua running on desktop would be a good candidate :) ). And make
sure the audio doesn't get routed through the server or otherwise this
wouldn't make any difference. You can call directly to the device's IP
address to make sure.
Also I tried the others codecs (iLBC, Speex/8000 and Speex/16000) in the
same way but didn't see any improvements.
iLBC and Speex/16000 is definitely out of question. Speex/8000 is probably
bang on the processing capability, so use it with care (e.g. only use
release mode), and probably is not good for troubleshooting problems like
this. And of course there is G.711, definitely a good candidate to try.
cheers
Benny
Do you have any suggestions or ideas?
We don't want to use Nokia APS (Audio proxy Server) for the moment, because
it needs a publisher ID.
Thank you,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Friday, March 20, 2009 2:32 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/19 George Evi george.evi@ctcinc.ca
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected Wi-Fi
and I expected to see some improvements but voice stilled disrupted.
I'm using iLBC as codec (1st priority) and UDP transport.
Aha, that's probably the reason. iLBC is heavy, I don't think the device has
enough processing power to run it [2]. Try with GSM or Speex.
Alternatively, consider using APS-Direct [1], available in pjsip version 1.1
now downloadable from the website. APS-Direct uses handset's native codec
and it supports iLBC, AMR, G.729, and G.711.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
[2] http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
2009/4/2 George Evi george.evi@ctcinc.ca
Hi Benny,
I activate the log in my project (PJ_LOG()) and used the function
“log_call_dump()*” *to dump the statistics at the end of a call.
I got these statistics:
Great! We love logs and statistics! :)
--end msg--State changed from Null to Calling, event=TX_MSGTransaction
tsx0x732e8c state changed to Calling
[CONFIRMED] To: sip:5148403000@sip6.van.netvoice.casip%3A5148403000@sip6.van.netvoice.ca
;tag=as5b69601c
Call time: 00h:01m:56s, 1st res in 3542 ms, conn in 5824ms
SRTP status: Not active Crypto-suite: (null)
#0 PCMU @8KHz, sendrecv, peer=64.34.49.82:19122
RX pt=0, stat last update: 00h:00m:00.601s ago
total 5.7Kpkt 924.8KB (1.15MB +IP hdr) @avg=62.3Kbps/77.9Kbps
pkt loss=143 (2.4%), discrd=1 (0.0%), dup=0 (0.0%), reord=1
(0.0%)
(msec) min avg max last dev
loss period: 20.000 178.750 940.000 100.000 63.638
jitter : - 0.001 11.737 579.000 0.750 8.775
TX pt=0, ptime=20ms, stat last update: 00h:00m:03.680s ago
total 5.7Kpkt 906.4KB (1.13MB +IP hdr) @avg 61.1Kbps/76.5Kbps
pkt loss=2 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 40.000 40.000 40.000 40.000 0.000
jitter : 193.000 193.000 193.000 193.000 0.000
RTT msec : 92.000 128.637 332.000 101.000 21.526
Processing incoming message: Response msg 200/BYE/cseq=18255
(rdata0x71e65c)RX 510 bytes Response msg 200/BYE/cseq=18255 (rdata0x71e65c)
from UDP 64.34.49.82:5060:
SIP/2.0 200 OK
I also read the “Understanding Media Flow” document and I have a (beginner)
question.
In the TX section we have a jitter line but in the Media Flow diagram there
is no Jitter Buffer for packet transmission, what represents this line?
We get these values from the RTCP report sent by the remote peer. If remote
peer doesn't support RTCP, we would not get these stats of course.
And also why in the same section the loss period and jitter buffer values
are the same for all statistics colons?
It's probably because it's only got one RTCP report? In this case then the
min/avg/max values would be the same, isn't it?
cheers
Benny
Thanks,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:
pjsip-bounces@lists.pjsip.org] *On Behalf Of *Benny Prijono
Sent: Wednesday, March 25, 2009 3:56 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/24 George Evi george.evi@ctcinc.ca
Hi Benny,
I update my application with the latest version the “trunk pjproject- 1.1”
and continue to test on Nokia E61.
As you suggested I changed the codec priorities in a way that GSM had
highest priority (in function “pjsua_media_subsys_init*” priority value
= PJMEDIA_CODEC_PRIO_NORMAL +4 (132)*). The sound was acceptable on
the caller side but on the callee side continue to be stuttered, disrupted
and instable.
Hi George,
In that case, I would probably suggest to try to use different peer for the
testing (pjsua running on desktop would be a good candidate :) ). And make
sure the audio doesn't get routed through the server or otherwise this
wouldn't make any difference. You can call directly to the device's IP
address to make sure.
Also I tried the others codecs (iLBC, Speex/8000 and Speex/16000) in the
same way but didn’t see any improvements.
iLBC and Speex/16000 is definitely out of question. Speex/8000 is
probably bang on the processing capability, so use it with care (e.g. only
use release mode), and probably is not good for troubleshooting problems
like this. And of course there is G.711, definitely a good candidate to try.
cheers
Benny
Do you have any suggestions or ideas?
We don’t want to use Nokia APS (Audio proxy Server) for the moment, because
it needs a publisher ID.
Thank you,
George.
From: pjsip-bounces@lists.pjsip.org [mailto:
pjsip-bounces@lists.pjsip.org] *On Behalf Of *Benny Prijono
Sent: Friday, March 20, 2009 2:32 AM
To: pjsip list
Subject: Re: [pjsip] Pjlib on Symbian
2009/3/19 George Evi george.evi@ctcinc.ca
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected
Wi-Fi and I expected to see some improvements but voice stilled disrupted.
I’m using iLBC as codec (1st priority) and UDP transport.
Aha, that's probably the reason. iLBC is heavy, I don't think the device
has enough processing power to run it [2]. Try with GSM or Speex.
Alternatively, consider using APS-Direct [1], available in pjsip version
1.1 now downloadable from the website. APS-Direct uses handset's native
codec and it supports iLBC, AMR, G.729, and G.711.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
[2] http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS
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