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[Freeswitch-users] PCMU fallback for T.38

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gkuri at ieee.org
Guest

Posted: Fri Mar 20, 2009 4:41 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

hey folks, I'm trying to configure PCMU fallback for T.38. The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with G729 and PCMU in the sdp. the INVITE to the provider includes G729 and PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... m=audio 16458 RTP/AVP 18 0 100 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 once the FAX tone is detected on the PSTN side, FS receives a T.38 re-INVITE from the provider and FS sends back a 488/Not Acceptable (proxy_media=false). at that point the provider than attempts fallback to PCMU with another reINVITE ... m=audio 16816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 immediately after the PCMU reINVITE, FS closes the channel and the text below is in the FS logs. given the SPA-2102 included PCMU in the original INVITE, even though it was the second preferred codec, shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by the provider? It seems like it's not passing the PCMU Re-INVITE back to the endpoint (SPA-2102), since it originally negotiated G729 with the SPA2102 as that was the 1st codec in the sdp, but trying to transcode between the two (G729 and PCMU)? 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 PCMU/8000 20 ms 160 samples 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1. 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 entering state [completed] 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Thanks, Gabe _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

brian at freeswitch.org
Guest

Posted: Fri Mar 20, 2009 4:42 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

Are you on SVN trunk 12694?

/b

On Mar 20, 2009, at 4:28 PM, Gabriel Kuri wrote:

Quote:

hey folks, I'm trying to configure PCMU fallback for T.38.

The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with
G729 and PCMU in the sdp. the INVITE to the provider includes G729 and
PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ...

m=audio 16458 RTP/AVP 18 0 100 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

once the FAX tone is detected on the PSTN side, FS receives a T.38
re-INVITE from the provider and FS sends back a 488/Not Acceptable
(proxy_media=false). at that point the provider than attempts fallback
to PCMU with another reINVITE ...

m=audio 16816 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

immediately after the PCMU reINVITE, FS closes the channel and the
text
below is in the FS logs. given the SPA-2102 included PCMU in the
original INVITE, even though it was the second preferred codec,
shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by
the provider? It seems like it's not passing the PCMU Re-INVITE back
to
the endpoint (SPA-2102), since it originally negotiated G729 with the
SPA2102 as that was the 1st codec in the sdp, but trying to transcode
between the two (G729 and PCMU)?

2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407
sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMU:0:8000]/[G729:18:8000]
2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371
sofia_glue_negotiate_sdp()
Set 2833 dtmf payload to 101
2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407
sofia_glue_negotiate_sdp()
Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000]
2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407
sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000]
2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550
sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU
2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601
sofia_glue_tech_set_codec() Set Codec
sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 PCMU/8000 20 ms 160
samples
2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811
sofia_glue_activate_rtp()
Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1
.
2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state()
Processing Reinvite
2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()
Channel sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 entering state
[completed]
2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655
switch_core_session_write_frame()
sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 receive message
[SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This
codec
is only usable in passthrough mode!
2009-03-20 01:19:58 [ERR] switch_core_io.c:723
switch_core_session_write_frame() Codec G.729 decoder error!

Thanks,

Gabe

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gkuri at ieee.org
Guest

Posted: Fri Mar 20, 2009 4:48 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

err, no, I tried upgrading from r11000 to r12669 yesterday, but starting
seeing crashing, so I have a jira open. currently I'm back on r11000.

http://jira.freeswitch.org/browse/FSCORE-338

Gabe

Brian West wrote:

Quote:

Are you on SVN trunk 12694?

/b

On Mar 20, 2009, at 4:28 PM, Gabriel Kuri wrote:

Quote:

brian at freeswitch.org
Guest

Posted: Fri Mar 20, 2009 4:55 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

Make current and try again... I haven't seen this crash you have seen... if you can run sippcapdump and get the packets that would help also.

Thanks,
/b

On Mar 20, 2009, at 4:41 PM, Gabriel Kuri wrote:

Quote:

err, no, I tried upgrading from r11000 to r12669 yesterday, but starting
seeing crashing, so I have a jira open. currently I'm back on r11000.

http://jira.freeswitch.org/browse/FSCORE-338

Gabe

gkuri at ieee.org
Guest

Posted: Fri Mar 20, 2009 6:13 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

OK, I'll give it a try and report back.

Gabe

Brian West wrote:

Quote:

Make current and try again... I haven't seen this crash you have seen...
if you can run sippcapdump and get the packets that would help also.

Thanks,
/b

On Mar 20, 2009, at 4:41 PM, Gabriel Kuri wrote:

Quote:

err, no, I tried upgrading from r11000 to r12669 yesterday, but starting
seeing crashing, so I have a jira open. currently I'm back on r11000.

http://jira.freeswitch.org/browse/FSCORE-338

Gabe

------------------------------------------------------------------------

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steveu at coppice.org
Guest

Posted: Fri Mar 20, 2009 9:53 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

Gabriel Kuri wrote:

Quote:

once the FAX tone is detected on the PSTN side, FS receives a T.38
re-INVITE from the provider and FS sends back a 488/Not Acceptable
(proxy_media=false). at that point the provider than attempts fallback
to PCMU with another reINVITE ...

This part is interesting, and the subject of a discussion we had
recently. A number of systems try that second re-invite after a 488, but
the SIP specs say the call is pretty much dead after the 488 message is
exchanged. Are they just hoping that maybe the other end will be
non-compliant enough to keep the call alive, and recover its media mode,
or haven't they read the specs?

Steve

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gkuri at ieee.org
Guest

Posted: Tue Mar 31, 2009 12:09 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

Quote:

I think they're hoping the other end is willing to recover it's media
mode rather than fail the call. I had no idea the call is technically
dead after the 488. Honestly, it would be nice if the call would still
be recoverable after that 488 on the T.38 ReINVITE, in order to try and
negotiate PCMU to try and keep the FAX going, but if that's not how it's
supposed to work, I'd rather follow the spec.

So for now I've disabled T.38 completely on both the SPA side and I had
the carrier disable it on my trunk, so they won't try a T.38 reinvite.
Instead they're trying a PCMU ReINVITE and the problem I'm seeing is
that if the carrier reINVITEs PCMU after the call initially started out
as G729, FS fails the call, because it seems to be trying to transcode
between G729 and PCMU, rather than pass the PCMU reINVITE through to the
other leg.

2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1550
sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU
2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1601
sofia_glue_tech_set_codec() Set Codec
sofia/cedarwireless.net/1909XXXXXXX@65.98.2
36.38 PCMU/8000 20 ms 160 samples
2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp()
Audio params are unchanged for sofia/cedarwireless.net/
1909XXXXXXX@65.98.236.38.
2009-03-31 00:30:50 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state()
Processing Reinvite
2009-03-31 00:30:50 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()
Channel sofia/cedarwireless.net/1909XXXXXXX@65.98.236.38 en
tering state [completed]
2009-03-31 00:30:50 [DEBUG] switch_core_io.c:655
switch_core_session_write_frame()
sofia/cedarwireless.net/1909XXXXXXX@65.98.23
6.38 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
2009-03-31 00:30:50 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-03-31 00:30:50 [ERR] switch_core_io.c:723
switch_core_session_write_frame() Codec G.729 decoder error!

I have inherit_codec=true, set in the dialplan and
disable-transcoding=true set in the sofia profile, which is what I
thought would do the trick, but it doesn't seem to be doing anything, FS
is still trying to transcode between G729 and PCMU. Is there something
I'm missing to get the PCMU ReINVITE from one of the legs to passthrough
to the other leg? Does this work only in proxy_media or bypass_media modes?

I am testing this with the latest rev of trunk as well.

Thanks,

Gabe

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anthony.minessale at g...
Guest

Posted: Tue Mar 31, 2009 2:37 pm Post subject: [Freeswitch-users] PCMU fallback for T.38

correct, we *do not* proxy re-invites except when bypass_media or proxy_media is set.

On Tue, Mar 31, 2009 at 11:59 AM, Gabriel Kuri <gkuri@ieee.org (gkuri@ieee.org)> wrote:

Quote:

> This part is interesting, and the subject of a discussion we had

Quote:

recently. A number of systems try that second re-invite after a 488, but
the SIP specs say the call is pretty much dead after the 488 message is
exchanged. Are they just hoping that maybe the other end will be
non-compliant enough to keep the call alive, and recover its media mode,
or haven't they read the specs?

I think they're hoping the other end is willing to recover it's media
mode rather than fail the call. I had no idea the call is technically
dead after the 488. Honestly, it would be nice if the call would still
be recoverable after that 488 on the T.38 ReINVITE, in order to try and
negotiate PCMU to try and keep the FAX going, but if that's not how it's
supposed to work, I'd rather follow the spec.

So for now I've disabled T.38 completely on both the SPA side and I had
the carrier disable it on my trunk, so they won't try a T.38 reinvite.
Instead they're trying a PCMU ReINVITE and the problem I'm seeing is
that if the carrier reINVITEs PCMU after the call initially started out
as G729, FS fails the call, because it seems to be trying to transcode
between G729 and PCMU, rather than pass the PCMU reINVITE through to the
other leg.

2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1550
sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU
2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1601
sofia_glue_tech_set_codec() Set Codec
sofia/cedarwireless.net/1909XXXXXXX@65.98.2
36.38 PCMU/8000 20 ms 160 samples
2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp()
Audio params are unchanged for sofia/cedarwireless.net/
1909XXXXXXX@65.98.236.38 (1909XXXXXXX@65.98.236.38).
2009-03-31 00:30:50 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state()
Processing Reinvite
2009-03-31 00:30:50 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()
Channel sofia/cedarwireless.net/1909XXXXXXX@65.98.236.38 en
tering state [completed]
2009-03-31 00:30:50 [DEBUG] switch_core_io.c:655
switch_core_session_write_frame()
sofia/cedarwireless.net/1909XXXXXXX@65.98.23
6.38 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
2009-03-31 00:30:50 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-03-31 00:30:50 [ERR] switch_core_io.c:723
switch_core_session_write_frame() Codec G.729 decoder error!

I have inherit_codec=true, set in the dialplan and
disable-transcoding=true set in the sofia profile, which is what I
thought would do the trick, but it doesn't seem to be doing anything, FS
is still trying to transcode between G729 and PCMU. Is there something
I'm missing to get the PCMU ReINVITE from one of the legs to passthrough
to the other leg? Does this work only in proxy_media or bypass_media modes?

I am testing this with the latest rev of trunk as well.

Thanks,

Gabe

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--
Anthony Minessale II

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AIM: anthm
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