Hi all, How do i get this to work:
From SIP client, this is what am sending: ================================
Content-Type: application/sdp Contact: <sip:1000@xx.xx.xx.xx.xx:6060;transport=UDP> v=0 o=user1 53655765 2353687637 IN IP4 xx.xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 1935 RTP/AVP 0 a=rtpmap:98 SPEEX/8000 (and, is 98 even the correct value here?) but freeswitc logs show this: ============================ ... 2010-11-23 15:54:19.947415 [DEBUG] sofia_glue.c:2741 Set Codec sofia/internal/1000@xx.xx.xx.xx PCMU/8000 20 ms 160 samples 64000 bits ..... 2010-11-23 15:54:19.993146 [DEBUG] mod_sofia.c:683 Local SDP sofia/internal/1000@xx.xx.xx.xx: v=0 o=FreeSWITCH 1290496453 1290496454 IN IP4 xx.xx.xx.xx s=FreeSWITCH c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 31206 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ..... Why is it sticking to PCMU/8000? and though RTP flows, audio not working. in vars.xml, i have: ... <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@8000h @20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@8000h @20i,PCMU,PCMA,GSM"/> ..... -- David Wafula
David, Your client is sending ulaw (the 0 on the m= line is ulaw) and speex on dynamic payload 98 which is fine. FreeSWITCH picks the first codec on your clients list and 200ok's if you wish to force your codec preferences over the clients you need to set codec-negotiation to greedy so that your in control on the server side and the client will not... currently you have the setting of generous I suspect. /b On Nov 23, 2010, at 10:22 AM, David Wafula wrote:
Hi all, How do i get this to work:
From SIP client, this is what am sending: ================================
Content-Type: application/sdp Contact: <sip:1000@xx.xx.xx.xx.xx:6060;transport=UDP>
v=0 o=user1 53655765 2353687637 IN IP4 xx.xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 1935 RTP/AVP 0 a=rtpmap:98 SPEEX/8000
(and, is 98 even the correct value here?)
but freeswitc logs show this: ============================ ... 2010-11-23 15:54:19.947415 [DEBUG] sofia_glue.c:2741 Set Codec sofia/internal/1000@xx.xx.xx.xx PCMU/8000 20 ms 160 samples 64000 bits .....
2010-11-23 15:54:19.993146 [DEBUG] mod_sofia.c:683 Local SDP sofia/internal/1000@xx.xx.xx.xx: v=0 o=FreeSWITCH 1290496453 1290496454 IN IP4 xx.xx.xx.xx s=FreeSWITCH c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 31206 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv .....
Why is it sticking to PCMU/8000?
and though RTP flows, audio not working. in vars.xml, i have:
... <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@8000h@20i,PCMU,PCMA,GSM"/> .....
Hi, i am not sure but i think your sip client is sending invalid SDP. it should look like this instead. m=audio 1935 RTP/AVP 0 a=rtpmap:0 SPEEX/8000 m=audio 1935 RTP/AVP 0
a=rtpmap:98 SPEEX/8000
Regards, -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com
BZZZ WRONG. The SDP is 100% valid. The SDP is saying ULAW payload 0 and Speex on payload 98. 0 without an rtpmap means ulaw. /b On Nov 23, 2010, at 10:50 AM, Ghulam Mustafa wrote:
Hi,
i am not sure but i think your sip client is sending invalid SDP. it should look like this instead.
m=audio 1935 RTP/AVP 0 a=rtpmap:0 SPEEX/8000
m=audio 1935 RTP/AVP 0 a=rtpmap:98 SPEEX/8000
Brian, Thanks for the correction Regards, -m On Tue, Nov 23, 2010 at 10:08 PM, Brian West <brian@freeswitch.org> wrote:
BZZZ WRONG.
The SDP is 100% valid. The SDP is saying ULAW payload 0 and Speex on payload 98. 0 without an rtpmap means ulaw.
/b
On Nov 23, 2010, at 10:50 AM, Ghulam Mustafa wrote:
Hi,
i am not sure but i think your sip client is sending invalid SDP. it should look like this instead.
m=audio 1935 RTP/AVP 0 a=rtpmap:0 SPEEX/8000
m=audio 1935 RTP/AVP 0
a=rtpmap:98 SPEEX/8000
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-- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com
May be my SIP knowledge is just too shallow, so this is what am getting: m=audio 1935 RTP/AVP 98 a=rtpmap:98 SPEEX/8000 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:115:32000:20:48000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:107:16000:20:32000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G722:9:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMU:0:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMA:8:8000:20:64000] 2010-11-23 19:54:29.302883 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[GSM:3:8000:20:13200] 2010-11-23 19:54:29.302883 [DEBUG] switch_channel.c:2457 (sofia/internal/1000@xx.xx.xx.xx) Callstate Change DOWN -> HANGUP 2010-11-23 19:54:29.302883 [NOTICE] sofia.c:4791 Hangup sofia/internal/1000@xx.xx.xx.xx [CS_NEW] [INCOMPATIBLE_DESTINATION] On Tue, Nov 23, 2010 at 7:23 PM, Ghulam Mustafa <mustafa.pk@gmail.com>wrote:
Brian,
Thanks for the correction
Regards, -m
On Tue, Nov 23, 2010 at 10:08 PM, Brian West <brian@freeswitch.org> wrote:
BZZZ WRONG.
The SDP is 100% valid. The SDP is saying ULAW payload 0 and Speex on payload 98. 0 without an rtpmap means ulaw.
/b
On Nov 23, 2010, at 10:50 AM, Ghulam Mustafa wrote:
Hi,
i am not sure but i think your sip client is sending invalid SDP. it should look like this instead.
m=audio 1935 RTP/AVP 0 a=rtpmap:0 SPEEX/8000
m=audio 1935 RTP/AVP 0
a=rtpmap:98 SPEEX/8000
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-- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com
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-- David Wafula
You possibly do not have mod_speex loaded or built. /b On Nov 23, 2010, at 2:02 PM, David Wafula wrote:
May be my SIP knowledge is just too shallow, so this is what am getting:
m=audio 1935 RTP/AVP 98 a=rtpmap:98 SPEEX/8000
2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:115:32000:20:48000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:107:16000:20:32000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G722:9:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMU:0:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMA:8:8000:20:64000] 2010-11-23 19:54:29.302883 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[GSM:3:8000:20:13200] 2010-11-23 19:54:29.302883 [DEBUG] switch_channel.c:2457 (sofia/internal/1000@xx.xx.xx.xx) Callstate Change DOWN -> HANGUP 2010-11-23 19:54:29.302883 [NOTICE] sofia.c:4791 Hangup sofia/internal/1000@xx.xx.xx.xx [CS_NEW] [INCOMPATIBLE_DESTINATION]
I can confirm mod_speex.so exists and is loaded in modules.conf.xml . am trying this again to see where am going wrong .. On Tue, Nov 23, 2010 at 10:40 PM, Brian West <brian@freeswitch.org> wrote:
You possibly do not have mod_speex loaded or built.
/b
On Nov 23, 2010, at 2:02 PM, David Wafula wrote:
May be my SIP knowledge is just too shallow, so this is what am getting:
m=audio 1935 RTP/AVP 98 a=rtpmap:98 SPEEX/8000
2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:115:32000:20:48000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:107:16000:20:32000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G722:9:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMU:0:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMA:8:8000:20:64000] 2010-11-23 19:54:29.302883 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[GSM:3:8000:20:13200] 2010-11-23 19:54:29.302883 [DEBUG] switch_channel.c:2457 ( sofia/internal/1000@xx.xx.xx.xx) Callstate Change DOWN -> HANGUP 2010-11-23 19:54:29.302883 [NOTICE] sofia.c:4791 Hangup sofia/internal/1000@xx.xx.xx.xx [CS_NEW] [INCOMPATIBLE_DESTINATION]
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-- David Wafula
Do you have it in your list of allowed codecs? -Steve On 24 November 2010 07:47, David Wafula <davidwaf@gmail.com> wrote:
I can confirm mod_speex.so exists and is loaded in modules.conf.xml . am trying this again to see where am going wrong ..
On Tue, Nov 23, 2010 at 10:40 PM, Brian West <brian@freeswitch.org> wrote:
You possibly do not have mod_speex loaded or built. /b On Nov 23, 2010, at 2:02 PM, David Wafula wrote:
May be my SIP knowledge is just too shallow, so this is what am getting: m=audio 1935 RTP/AVP 98 a=rtpmap:98 SPEEX/8000 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:115:32000:20:48000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:107:16000:20:32000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G722:9:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMU:0:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMA:8:8000:20:64000] 2010-11-23 19:54:29.302883 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[GSM:3:8000:20:13200] 2010-11-23 19:54:29.302883 [DEBUG] switch_channel.c:2457 (sofia/internal/1000@xx.xx.xx.xx) Callstate Change DOWN -> HANGUP 2010-11-23 19:54:29.302883 [NOTICE] sofia.c:4791 Hangup sofia/internal/1000@xx.xx.xx.xx [CS_NEW] [INCOMPATIBLE_DESTINATION]
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-- David Wafula
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Checked again, ya, that was it. Had accidentally commented out speex in allowed codecs. Thanks. On Wed, Nov 24, 2010 at 11:13 AM, Steven Ayre <steveayre@gmail.com> wrote:
Do you have it in your list of allowed codecs?
-Steve
On 24 November 2010 07:47, David Wafula <davidwaf@gmail.com> wrote:
I can confirm mod_speex.so exists and is loaded in modules.conf.xml . am trying this again to see where am going wrong ..
On Tue, Nov 23, 2010 at 10:40 PM, Brian West <brian@freeswitch.org> wrote:
You possibly do not have mod_speex loaded or built. /b On Nov 23, 2010, at 2:02 PM, David Wafula wrote:
May be my SIP knowledge is just too shallow, so this is what am getting: m=audio 1935 RTP/AVP 98 a=rtpmap:98 SPEEX/8000 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:115:32000:20:48000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G7221:107:16000:20:32000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[G722:9:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMU:0:8000:20:64000] 2010-11-23 19:54:29.301838 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[PCMA:8:8000:20:64000] 2010-11-23 19:54:29.302883 [DEBUG] sofia_glue.c:4367 Audio Codec Compare [SPEEX:98:8000:0:0]/[GSM:3:8000:20:13200] 2010-11-23 19:54:29.302883 [DEBUG] switch_channel.c:2457 (sofia/internal/1000@xx.xx.xx.xx) Callstate Change DOWN -> HANGUP 2010-11-23 19:54:29.302883 [NOTICE] sofia.c:4791 Hangup sofia/internal/1000@xx.xx.xx.xx [CS_NEW]
[INCOMPATIBLE_DESTINATION]
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-- David Wafula
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-- David Wafula
participants (4)
-
Brian West
-
David Wafula
-
Ghulam Mustafa
-
Steven Ayre