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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