Hi everybody, I'm stuck trying to make pjsip 1.8 work with my project, so I
finally decide to ask for help, I usually read the mails I receive from the
pjsip mailing list and I can see that a lof of you guys have very good
experience, so I some of you can point me in the right direction.
My problem is this:
I have mi VOIP app for iPhone using pjsip 1.0 working really nice, so I
decide to migrate to pjsip 1.8 to add multitasking support and receiving
calls while de iPhone is locked.
I downloaded the code and compile it using ./configure-iphone added the
libraries to my project and run it, everything fine but Im unable to hear
voice in local speaker, the guy in the other side can hear me pretty well so
I decide to follow the Troubleshooting sound problems section in the pjsip
site using the pjsua app.
I run it, add new account with +a, registered successfully with SIP server
and made a call with the same problem.
These are my results for the checklist suggestions:
1. Check that the correct device is being used.
Here i see:
*pjsua_media.c Opening sound device PCM at 8000/1/20ms
while making the call and
pjsua_media.c
Closing sound device after idle for 1 seconds
pjsua_media.c Closing iPhone IO device sound playback device and iPhone
IO device sound capture device
while hanging up the call.
*
2. Check that no other application is using the devices. These isn't the
case, no other audio is been played.
3. Check that speaker is functioning properly by Looping-back Microphone
to Speaker device. The speaker is working properly because if change de
include folder path to the one holding the pjsip 1.0 version everything
works fine.
4. You can also check by Playing a WAV File to Speaker. I haven't tried
this but I think it should work for the same reason that checklist number 3,
but I always set the Volume level to max.
5. Check that Incoming RTP Packets are Indeed Received by Local
Host. *Indeed
no RTP packets are been received *this is what I saw in the log while
using dq command:
[CONFIRMED] To: sip:349201031 at abc.domain.com<sip%3A349201031 at abc.domain.com>
;tag=lignup-3896-4ca3a0a3
Call time: 00h:01m:09s, 1st res in 4715 ms, conn in 6238ms
SRTP status: Not active Crypto-suite: (null)
#0 PCMA @8KHz, sendrecv, peer=-
RX pt=8, stat last update: 00h:00m:02.427s ago
total 1pkt 0B (40B +IP hdr) @avg=0bps/4bps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0
(0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
TX pt=8, ptime=20ms, stat last update: never
total 3.5Kpkt 570.2KB (712.8KB +IP hdr) @avg 63.9Kbps/79.8Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
But I don't know what exactly I can do here, I have tried to disable VAD
I dont think NAT could be the problem, Im using 3G and I have tried in
several Wi-Fi to test if it was a router problem, besides that it works fine
in the same networks with pjsip 1.0 and the same SIP servers accounts.
By the other hand, when I try to use the --use-stun1 or --no-vad from
command line in pjsua it always says "Invalid input" don't know why?
6. Getting Around NAT Explained above in checklist #5
7. Check that the Call is Connected to the Sound Device in the Conference
Bridge. When i use cl to see the port list I see:
Conference ports:
Port #00[ 8KHz/20ms/1] iPhone IO device transmitting to: #3
Port #01[ 8KHz/20ms/1] ringback transmitting to:
Port #02[ 8KHz/20ms/1] ring transmitting to:
Port #03[ 8KHz/20ms/1] sip: 349201031 at abc.domain.com transmitting to: #0
So, apparently ports are connected properly ant transmitting.
8. Check that CPU Utilization is not Too High This is not the case, is
the only app running in the iPhone.
I tried to put all the information I could to help everybody to understand
better my problem and I can add more information if needed/requested.
I even tried to add multitasking and running on background support to pjsip
1.0 but even I was able to receive calls on background and use other apps
while in a call with or without speaker, whenever the apps get locked I stop
receiving calls and when I tried to return to my app I receive a SIGPIPE
error that I haven't been enable to fix.
I really think that making work the pjsip 1.8 is a better approach and I
feel I'm just missing an easy step to make it work, so I hope someone here
could take the time to help me.
By the way: I also been able to receive incoming calls on pjsua from pjsip
1.8, I dont know if this is related with the fact that Im not receiving rtp
packets when on call, and this is another problem I can try to fix by
miself, just wanted to note in case is related with this problem.
Thanks in advance for your time!
Regards,
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