Hi,
first of all, please use 0.8.0 for Symbian. It has many improvements
over 0.7.0, especially for Symbian.
Regarding your link errors, we discussed this in the other thread
("Sip stack 8.0 PJMedia have link errors"), could you please follow
up my suggestion in that thread?
Then regarding the sound device, there are two sound "devices"
implementation for Symbian: null_audio.mmp (the name says it all),
and symbian_audio.mmp for the real sound device. Please use
symbian_audio.mmp if you want audio. But mind you, this
implementation is pretty much experimental, so there is a big chance
that it will not work.
regards,
-benny
saurabh agarwal wrote:
> Hi,
>
> I am trying to implement CALL functionality in my application which i
> have developed on Symbian S60 FP1 3rd edition. I am using Code Warrior
> as my IDE. First there are many link errors when i m compiling and
> linking PJSip Stack available on your site. There are many link errors
> in version 0.8 and also in SVN version.
> When I m using version 0.7 its compliling and technically i am able to
> achieve CALL functiobnality that is I am getting all the required
> responses from server like user being registered, contact header,
> dialogue are created properly and i get 180 ringing and 200 ok as
> response from server but not able to transmit voice across. Please help
> me out. Few of the linking errors which i got while compiling version
> 0.8 are shown below:
>
> /tool exit status == 1
> />/ Errors caused tool to abort.
> />/
> />/ Link Error : Undefined symbol: '_pjmedia_snd_stream_get_info'
> />/ referenced from '_create_sound_port' in conference.c:451
> />/ referenced from '_pjmedia_snd_port_set_ec' in sound_port.c:481
> />/ Link Error : Undefined symbol: '_pjmedia_snd_get_dev_info'
> />/ referenced from '_create_sound_port' in conference.c:455
> />/ referenced from '_create_sound_port' in conference.c:457
> />/ Link Error : Undefined symbol: '_speex_aec_cancel_echo'
> />/ referenced from '.data' in echo_common.o
