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tbskyd at gmail.com
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 Posted: Sat Sep 27, 2014 10:28 am Post subject: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC? |
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hi:
when using chan_sip, I can use set SIP_CODEC in dialplan to change
the codec of endpoint. this method didn't work with pjsip in asterisk
12/13.
I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER.
according to the description, it seems can set codec, but the document
didn't offer any example. i try to use something like
PJSIP_MEDIA_OFFER(alaw) but didn't work.
can someone give an example for the function? thanks for the help.
Regards,
tbskyd
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universe at truemetal.org
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| Posted: Sun Sep 28, 2014 1:01 am Post subject: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC? |
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Am 27.09.2014 17:28, schrieb d tbsky:
| Quote: | can someone give an example for the function? thanks for the help.
|
Not a programmer here, just grep -r'ed through the code, but maybe try
one of these:
G711A
G711_ALAW
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tbskyd at gmail.com
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| Posted: Sun Sep 28, 2014 10:08 am Post subject: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC? |
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2014-09-28 14:01 GMT+08:00 Markus <universe@truemetal.org>:
| Quote: | Am 27.09.2014 17:28, schrieb d tbsky:
| Quote: |
can someone give an example for the function? thanks for the help.
|
Not a programmer here, just grep -r'ed through the code, but maybe try one
of these:
G711A
G711_ALAW
|
thanks a lot for help!! I tried both but none works. maybe this
function can not work like the old channel variable "SIP_CODEC", which
can change inbound call codec. but I do notice something different
between chan_sip and chan_pjsip.
I use zoiper softphone for testing:
when I dialout sip trunk with chan_sip, the remote peer rings, and
zoiper now shows what codec to use. if I use "SIP_CODEC" before dial
to change the codec, zoiper will use the new CODEC, but asterisk
internal won't change and still transcoding in the middle.(at least
"core show channel sip/xxxxx" told me transcoding)
when I dialout sip trunk with chan_pjsip, the remote peer rings, but
zoiper didn't show what codec to use. only after the callee answer the
phone, zoiper shows what codec to use. so it seems chan_pjsip have
better chance to do the right thing without transcoding. it's sad that
chan_pjsip won't select best codec match two peers automatically
without transcoding. but I hope it at least can provide a magic
function or channel variable like "SIP_CODEC/SIP_CODEC_INBOUND" to
make correct codec selection.
Regards,
tbskyd
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mjordan at digium.com
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| Posted: Tue Sep 30, 2014 10:53 am Post subject: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC? |
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On Sat, Sep 27, 2014 at 10:28 AM, d tbsky <tbskyd@gmail.com> wrote:
| Quote: | hi:
when using chan_sip, I can use set SIP_CODEC in dialplan to change
the codec of endpoint. this method didn't work with pjsip in asterisk
12/13.
I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER.
according to the description, it seems can set codec, but the document
didn't offer any example. i try to use something like
PJSIP_MEDIA_OFFER(alaw) but didn't work.
can someone give an example for the function? thanks for the help.
|
The function should work on whatever channel it was set on. If you are
going to use it on an outbound channel, then you should use a pre-dial
handler to apply it to that channel.
Matt
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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tbskyd at gmail.com
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| Posted: Tue Sep 30, 2014 12:48 pm Post subject: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC? |
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2014-09-30 23:52 GMT+08:00 Matthew Jordan <mjordan@digium.com>:
| Quote: | On Sat, Sep 27, 2014 at 10:28 AM, d tbsky <tbskyd@gmail.com> wrote:
| Quote: | I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER.
according to the description, it seems can set codec, but the document
didn't offer any example. i try to use something like
PJSIP_MEDIA_OFFER(alaw) but didn't work.
can someone give an example for the function? thanks for the help.
|
The function should work on whatever channel it was set on. If you are
going to use it on an outbound channel, then you should use a pre-dial
handler to apply it to that channel.
|
it sounds good. could you give out an one line dialplan example so I
can try to use it? and the real thing I want to change is the inbound
codec, can it work like the chan_sip channel variable
SIP_CODEC_INBOUND?
thanks a lot for your help!!
Regards,
tbskyd
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