Asterisk canreinvite (directmedia) equivalent in freeswitch
Does freeswitch have anything similar to canreinvite (asterisk sip setting)? I know there is bypass_media but it does not work in the same way as canreinvite in asterisk. It looks like bypass_media always tells freeswitch to send media directly between originator and terminator even though they do not have common codecs. I want to use bypass_media only when leg A and leg B have common codecs. If they do not have then freeswitch should stay in media path and handle codec transcoding. Is there an easy way to achieve this in freeswitch?
Theres a few ways to accomplish this... But you are probably looking for something like bypass_after_bridge K On 6/30/15, 12:16 PM, "Ričardas Stoma" <ricardas.stoma@gmail.com> wrote:
Does freeswitch have anything similar to canreinvite (asterisk sip setting)? I know there is bypass_media but it does not work in the same way as canreinvite in asterisk.
It looks like bypass_media always tells freeswitch to send media directly between originator and terminator even though they do not have common codecs.
I want to use bypass_media only when leg A and leg B have common codecs. If they do not have then freeswitch should stay in media path and handle codec transcoding.
Is there an easy way to achieve this in freeswitch?
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I tried bypass_media_after_bridge but calls still get connected with different codecs and all i can hear is digital noise. 2015-07-01 15:34 GMT+03:00 Ken Rice <krice@freeswitch.org>:
Theres a few ways to accomplish this... But you are probably looking for something like bypass_after_bridge
K
On 6/30/15, 12:16 PM, "Ričardas Stoma" <ricardas.stoma@gmail.com> wrote:
Does freeswitch have anything similar to canreinvite (asterisk sip setting)? I know there is bypass_media but it does not work in the same way as canreinvite in asterisk.
It looks like bypass_media always tells freeswitch to send media directly between originator and terminator even though they do not have common codecs.
I want to use bypass_media only when leg A and leg B have common codecs. If they do not have then freeswitch should stay in media path and handle codec transcoding.
Is there an easy way to achieve this in freeswitch?
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If you are hearing noise then you have a different problem... Post a full trace of the call... Sounds more like a broken endpoint. And don't forget you probably want to do some things with the SDPs to make sure you have matching codecs... You can do testing against them in the dialplan and other ways On 7/1/15, 8:06 AM, "Ričardas Stoma" <ricardas.stoma@gmail.com> wrote:
I tried bypass_media_after_bridge but calls still get connected with different codecs and all i can hear is digital noise.
2015-07-01 15:34 GMT+03:00 Ken Rice <krice@freeswitch.org>:
Theres a few ways to accomplish this... But you are probably looking for something like bypass_after_bridge
K
On 6/30/15, 12:16 PM, "Ričardas Stoma" <ricardas.stoma@gmail.com <http://ricardas.stoma@gmail.com> > wrote:
Does freeswitch have anything similar to canreinvite (asterisk sip setting)? I know there is bypass_media but it does not work in the same way as canreinvite in asterisk.
It looks like bypass_media always tells freeswitch to send media directly between originator and terminator even though they do not have common codecs.
I want to use bypass_media only when leg A and leg B have common codecs. If they do not have then freeswitch should stay in media path and handle codec transcoding.
Is there an easy way to achieve this in freeswitch?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <http://consulting@freeswitch.org> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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I think everything is ok with my endpoints. I use two softphones (linphone and zoiper). On linphone (leg A) i enable only GSM codec and on zoiper (leg B) i enable only PCMA codec. When bypass_media_after bridge is disabled, media goes through freeswitch and freeswitch does transcoding. In this case i can hear voices. When bypass_media_after_bridge is enabled, media goes from leg A to leg B directly but the problem is that codecs do not match and i can hear only digital noise. SIP trace looks like this http://i.imgur.com/iOn6ajJ.png If i set PCMA codec on linphone (leg A) then everything is ok with direct media between softphones, codecs matches and i can hear voices. What i want is IF leg A and leg B codecs matches then bypass_media should take effect and media should go directly from leg A to leg B. IF codecs do not match, then media should go through freeswitch and freeswitch should do transcoding. Is this possible with freeswitch? Asterisk does this when canreinvite is enabled ("If the clients use different codecs, Asterisk will not issue a re-invite")
For bypass to work you'll have to have some codecs in common! Without that it sounds like your end points don't behave properly in his case. /b On Friday, July 3, 2015, Ričardas Stoma <ricardas.stoma@gmail.com> wrote:
I think everything is ok with my endpoints. I use two softphones (linphone and zoiper). On linphone (leg A) i enable only GSM codec and on zoiper (leg B) i enable only PCMA codec. When bypass_media_after bridge is disabled, media goes through freeswitch and freeswitch does transcoding. In this case i can hear voices.
When bypass_media_after_bridge is enabled, media goes from leg A to leg B directly but the problem is that codecs do not match and i can hear only digital noise. SIP trace looks like this http://i.imgur.com/iOn6ajJ.png
If i set PCMA codec on linphone (leg A) then everything is ok with direct media between softphones, codecs matches and i can hear voices. What i want is IF leg A and leg B codecs matches then bypass_media should take effect and media should go directly from leg A to leg B. IF codecs do not match, then media should go through freeswitch and freeswitch should do transcoding.
Is this possible with freeswitch? Asterisk does this when canreinvite is enabled ("If the clients use different codecs, Asterisk will not issue a re-invite")
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Yes, i see that bypass_media works only if end points have common codecs. That's why i'm asking if there is a way to use bypass_media only when end points have common codecs, otherwise freeswitch should handle transcoding. Asterisk has single setting called 'canreinvite' which works this way. Maybe freeswitch has something like this? Or does it require some fancy dialplan/scripting to achieve this? 2015-07-03 17:05 GMT+03:00 Brian West <brian@freeswitch.org>:
For bypass to work you'll have to have some codecs in common! Without that it sounds like your end points don't behave properly in his case.
/b
On Friday, July 3, 2015, Ričardas Stoma <ricardas.stoma@gmail.com> wrote:
I think everything is ok with my endpoints. I use two softphones (linphone and zoiper). On linphone (leg A) i enable only GSM codec and on zoiper (leg B) i enable only PCMA codec. When bypass_media_after bridge is disabled, media goes through freeswitch and freeswitch does transcoding. In this case i can hear voices.
When bypass_media_after_bridge is enabled, media goes from leg A to leg B directly but the problem is that codecs do not match and i can hear only digital noise. SIP trace looks like this http://i.imgur.com/iOn6ajJ.png
If i set PCMA codec on linphone (leg A) then everything is ok with direct media between softphones, codecs matches and i can hear voices. What i want is IF leg A and leg B codecs matches then bypass_media should take effect and media should go directly from leg A to leg B. IF codecs do not match, then media should go through freeswitch and freeswitch should do transcoding.
Is this possible with freeswitch? Asterisk does this when canreinvite is enabled ("If the clients use different codecs, Asterisk will not issue a re-invite")
--
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*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
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*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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participants (3)
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Brian West
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Ken Rice
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Ričardas Stoma