View Single Post
Posts: 290 | Thanked: 385 times | Joined on Jan 2012 @ Madrid, Spain
#70
Originally Posted by TheGrave View Post
I've been busting my head for a few days trying to figure out how to make telepathy-rakia to see all gstreamer codecs. My particular interest is G.729. Do you have any clue whether some config file has to be modified? farstream seems to have it enabled but Wireshark says it's not offered by the phone:

cat /usr/share/farstream/0.1/fsrtpconference/default-codec-preferences
################
# Audio codecs #
################

[audio/SPEEX:8000]
clock-rate=8000

[audio/SPEEX:16000]
clock-rate=16000

[audio/AMR]

[audio/G729]

[audio/ILBC]

Codec seems to be installed:

gst-inspect-0.10 | grep 729
rtp: rtpg729depay: RTP G.729 depayloader
rtp: rtpg729pay: RTP G.729 payloader
Hi.
I'm struggling to make VOIP calls work. I'm not able to make the sip client offer any codecs:
Code:
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 124

v=0
o=- 2847608483721697075 2098627415485425572 IN IP4 172.26.0.16
s=-
c=IN IP4 172.26.0.16
t=0 0
m=audio 0 RTP/AVP 0
, and my SIP server drops the call just when I answer. I followed instrucctions in the Jolla Forum, but when I try to restart pulseaudio service I always get a "file not found" error.
These are the codecs offered by my Asterisk PBX:
Code:
Content-Type: application/sdp
Content-Length: 339

v=0
o=root 1205889408 1205889408 IN IP4 172.26.0.2
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 172.26.0.2
t=0 0
m=audio 14532 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Am I missing something? This is my 2nd day tinkering with the device, BTW
Regards.