Yeah, thank you guys, i didn't know that those lines 'a=rtpmap' are optional. Everything is working as it should now ;-) 2014-06-06 23:46 GMT+03:00 Michael Collins <msc@freeswitch.org>:
Well, it seems my mail thread management has failed me. I totally missed KK's post here. At least we gave him the same info. :P -MC
On Fri, Jun 6, 2014 at 11:15 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
The codecs were included from the start (except GSM):
m=audio 17262 RTP/AVP 8 0 101 13
8 = PCMA 0 = PCMU 101 = telephone-event (dynamic) 13 = CNG
See this:
https://wiki.freeswitch.org/wiki/Variable_verbose_sdp
You might not be seeing GSM because it's not included on the profile or the module (mod_spandsp, I believe) isn't loaded.
On Fri, Jun 6, 2014 at 9:51 AM, Ričardas Stoma <ricardas.stoma@gmail.com> wrote:
It is still the same without single quote marks around PCMA,PCMU,GSM...
But i think everything is OK with my SDP, i just made a call to my asterisk server and i see that asterisk receives correct codecs:
Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
So i guess i had to learn more about SDP and codecs before posting my issue, sorry guys :-)
2014-06-06 15:47 GMT+03:00 Anonio Silva <asilva@wirelessmundi.com>:
Can you try:
{absolute_codec_string=PCMA,PCMU,GSM} ===> without '
On Fri, 2014-06-06 at 11:36 +0300, Ričardas Stoma wrote:
I have created the following extension:
<extension name="my_test"> <condition> <action application="bridge" data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/ 1001@192.168.1.78"/> <action application="hangup"/> </condition> </extension>
then i call to this extension, freeswitch dials to 1001@192.168.1.78 and when i check the content of SDP, included in a SIP packet, i see something like this:
v=0 o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65 s=FreeSWITCH c=IN IP4 192.168.1.65 t=0 0 m=audio 17262 RTP/AVP 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
why my codecs from 'absolute_codec_string' are not included in SDP? I would expect to see these lines included:
a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000
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_________________________________________________________________________
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