[Asterisk-Users] linphone -> NAT -> * -> NAT -> firefly woes.

[Bruno Hertz]

On Wed, 2005-01-12 at 14:39 -0800, Erik Espinoza wrote:
> Did you enable passthrough for the rtp ports on the asterisk box?
> 
> I had the same problem until I enabled udp 10000:20000 on the firewall.

I did. That's why linphone -> * echo test works.

Maybe I made some progress however, by logging linphone output and
comparing the successful echo test to the unsuccessful iax bridge.

On echo test I see:

(linphone:5450): LinphoneCore-WARNING **: payload PCMA is not usable or enabled.
(linphone:5450): LinphoneCore-WARNING **: This remote sip phone did not answer properly to my sdp offer!
MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> GSMEncoder,0
MediaStreamer-Message: ms_filter_add_link: GSMEncoder,0 -> RTPSend,0
MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> GSMDecoder,0
MediaStreamer-Message: ms_filter_add_link: GSMDecoder,0 -> OssWrite,0
MediaStreamer-Message: Opening sound card in capture mode with stereo=0,rate=8000,bits=16
MediaStreamer-Message: dsp blocksize is 512.
MediaStreamer-Message: Opening sound card in playback mode with stereo=0,rate=8000,bits=16

whereas on linphone -> * -> firefly:

(linphone:5456): LinphoneCore-WARNING **: payload PCMA is not usable or enabled.
(linphone:5456): LinphoneCore-WARNING **: This remote sip phone did not answer properly to my sdp offer!
MediaStreamer-Message: Mediastreamer processing thread is exiting.

I.e. on echo test linphone does select the gsm codec, while with
iax bridge the media stream is canceled immediately, hence it stops
sending data as properly reported by *. Maybe it's a codec issue,
I'm just in the process of investigating ... just thinking, doesn't
* transcode between channel legs if necessary, could it be I disabled
that by accident (?) ...

Thanks, Bruno.