Show Posts

This section allows you to view all posts made by this member. Note that you can only see posts made in areas you currently have access to.

Topics - looserouting

Pages: [1]
Yate bugs / iLBC negotiation
« on: June 04, 2014, 07:55:21 AM »
when yate offers iLBC mode=20 and the remote UA anwers with mode=30 then yate must use mode=30. But it doesn't....

AFAIK iLBC mode=20 must only be used if both prefer mode=20.

I am using regfile and regexroute for routing.
regfile first then regexroute.
the users are all SIP-users.

i am using regfile for routing to local users and regexroute to call out through my voip provider(configured in accfile).
now if i call a user and he is registered then regfile will route the call.
but if he's not the call will be routed through regexroute.

how can I send an 480 (or user absent for other channels then sip) if the user is offline ?

if i don't load regexroute then yate will send a 480 response. but then i can't call out :)

Yate users hangout place / Mapping of Response codes SIP<-->ISUP
« on: July 30, 2013, 07:33:05 AM »
is yate compliant with Q.1912.5 ?
Can Q.850 Reasons be inserted in SIP headers ?
If not, are there any plans ?
If yes, where can i find it in the code ?

Yate users hangout place / Please help me with my tests
« on: July 22, 2013, 09:37:40 AM »
As RFC4566 told us
The purpose of SDP is to convey information about media streams in
   multimedia sessions to allow the recipients of a session description
   to participate in the session

So if an UAC (alice )sends an SIP-INVITE with let's say g711u and an packetization time of 70ms the receiver (Bob)should only accept the call if it can handle that ptime and should also use the same. Otherwise the INVITE should be declined with 488 Not Acceptable.

But many UAC's (Bob's) whould answer this call with ptime 20 (or its default).  A good UAC can react and use the ptime in Bob's 200 OK. Others don't , so both Endpoint use diffret packetization times.

Now I wanted to check out how yate behave in the middle of these szenarios.

My questions for now are:
How can I change the ptime in regexroute or routing module ?
How can I make yate respect the ptime send by the caller ? (i uses alway its default)

do i have to "subscribe" the chan.disconnect message (like in extmodule )or do i always receive it ?
how can i check which message i 've got ?
sady there aren't much examples for this module

Yate users hangout place / Transcode encrypted Streams
« on: May 24, 2013, 09:42:39 AM »
I enabled SRTP on Caller side and did a call.

yate sends SDP in SIP  to callee with "m=audio 25334 RTP/SAVP 8 9 101"  but no crypto field.

Is SRTP only possible if yate stays out of the media path ?

Yate users hangout place / Transcoding and ptime
« on: May 24, 2013, 07:17:56 AM »
when i transcode can I "offer" ilbc30 and g711 with ptime:20 ?
afaik yate sets the ptime equal to ilbc mode.
Where can I set the default ptime for other codecs then ilbc (or codecs with its own ptime settings like ilbc)?

another thing i saw: if I just enable ilbc (not ilbc20 or ilbc30) in ysipchan.conf then ilbc will not be used at all.

Pages: [1]