VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
jeff at jeff.net
Guest
|
 Posted: Tue Mar 24, 2015 4:15 pm Post subject: [asterisk-users] RTP handling |
 |
|
Hello,
I am wondering if asterisk does anything at all to RTP packets passed
from channel to channel if no transcoding is involved? Can I assume that
the packet that left phone A, arrived at the asterisk server, was copied
to phone B's channel and eventually arrived at phone B had exactly (byte
for byte) the same payload? Assume two SIP endpoints, no NAT involved.
Thanks,
j
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
 |
rmudgett at digium.com
Guest
|
| Posted: Tue Mar 24, 2015 4:28 pm Post subject: [asterisk-users] RTP handling |
|
|
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff@jeff.net (jeff@jeff.net)> wrote:
| Quote: |
Hello,
I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved.
|
That will only happen when the call is natively bridged:
Non-native bridge: Packets can get translated or Asterisk has an interest in the packet for things like DTMF or call recording.
Native bridge doing packet-to-packet (Local bridging): Packets come in on one channel and go out the other channel with nothing else done to them.
Native bridge doing direct media (Remote bridging): Packets go directly between endpoints so Asterisk never sees them.
Richard |
|
| Back to top |
|
|
jeff at jeff.net
Guest
|
| Posted: Tue Mar 24, 2015 4:57 pm Post subject: [asterisk-users] RTP handling |
|
|
On 03/24/2015 04:28 PM, Richard Mudgett wrote:
| Quote: |
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff@jeff.net (jeff@jeff.net)> wrote:
| Quote: |
Hello,
I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved.
|
That will only happen when the call is natively bridged:
Non-native bridge: Packets can get translated or Asterisk has an interest in the packet for things like DTMF or call recording.
Native bridge doing packet-to-packet (Local bridging): Packets come in on one channel and go out the other channel with nothing else done to them.
Native bridge doing direct media (Remote bridging): Packets go directly between endpoints so Asterisk never sees them.
Richard
|
Thanks for the quick reply RIchard! Can I force native bridging, or does it default to that if I don't configure direct media? The dialplan will be very simple - extensions calling extensions within a context. No DTMF, no recording, no mixing for conference, etc.
Cheers,
j |
|
| Back to top |
|
|
rmudgett at digium.com
Guest
|
| Posted: Tue Mar 24, 2015 5:07 pm Post subject: [asterisk-users] RTP handling |
|
|
On Tue, Mar 24, 2015 at 4:59 PM, Jeff LaCoursiere <jeff@jeff.net (jeff@jeff.net)> wrote:
| Quote: | On 03/24/2015 04:28 PM, Richard Mudgett wrote:
| Quote: |
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff@jeff.net (jeff@jeff.net)> wrote:
| Quote: |
Hello,
I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved.
|
That will only happen when the call is natively bridged:
Non-native bridge: Packets can get translated or Asterisk has an interest in the packet for things like DTMF or call recording.
Native bridge doing packet-to-packet (Local bridging): Packets come in on one channel and go out the other channel with nothing else done to them.
Native bridge doing direct media (Remote bridging): Packets go directly between endpoints so Asterisk never sees them.
Richard
|
Thanks for the quick reply RIchard! Can I force native bridging, or does it default to that if I don't configure direct media? The dialplan will be very simple - extensions calling extensions within a context. No DTMF, no recording, no mixing for conference, etc.
|
You cannot force native bridging. It will switch to native bridging if you don't set anything
that makes Asterisk interested in the media stream. Such as enabling
DTMF features in features.conf and Dial flags like t or T.
Richard |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
