Codec Negotiation: Codec telephone-event:101:8000:20
Hi! If I try to call out on one of my voip providers I get INCOMPATIBLE_DESTINATION. Something is going wrong with codec negotiation: 2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3262 Set 2833 dtmf payload to 101 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[PCMU:0:8000:20] 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [CN:13:8000:20]/[PCMA:8:8000:20] 2010-01-05 18:57:56.845029 [NOTICE] sofia.c:3937 Hangup sofia/external/020555500 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] According to their website, the codec they us is G711 a-law or G729. What codecs have name CN and telephone-event? How do I get these to match? I assume the format of the debug output is: Audio Codec compare [Codec Name:Media Format:Media Name:Rate:RTP Size?] Thanks
your invite says G726-32 (thats what the 2 is in the audio line) /b On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I thought the Remote SDP comes from my SIP provider? o=Sippy 257534956 1 IN IP4 80.232.37.178. 80.232.37.178 is not my IP address, its the ip address of my voip provider? But now you mention that 2=G726-32, its what I have as default. Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why isn't negotiation used to select these codecs and isn't FS comparing codecs? Sorry, I guess these are basic questions ... On Wed, Jan 6, 2010 at 2:45 PM, Brian West <brian@freeswitch.org> wrote:
your invite says G726-32 (thats what the 2 is in the audio line)
/b
On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
_______________________________________________ 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
because they aren't in the invite... it can't negotiate things that aren't in the invite... its clearly NOT in that sdp... its only CN, G726-32 and Telephony event. /b On Jan 5, 2010, at 9:59 PM, Mark Campbell-Smith wrote:
Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why isn't negotiation used to select these codecs and isn't FS comparing codecs?
Sorry, I guess these are basic questions ...
For my codec prefs I have: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=iLBC@30i,G726-32,G722,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G726-32,PCMU,PCMA"/> How many codecs are sent in an invite? Is it only the top three of global_codec_prefs? Is CN = iLBC and telephony-event = G722? Thanks On Wed, Jan 6, 2010 at 3:02 PM, Brian West <brian@freeswitch.org> wrote:
because they aren't in the invite... it can't negotiate things that aren't in the invite... its clearly NOT in that sdp... its only CN, G726-32 and Telephony event.
/b
On Jan 5, 2010, at 9:59 PM, Mark Campbell-Smith wrote:
Anyway, even if G726-32 is default, I also have PCMU and PCMA ... why isn't negotiation used to select these codecs and isn't FS comparing codecs?
Sorry, I guess these are basic questions ...
_______________________________________________ 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
Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden. /b On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
ahh... thats what I thought originally. So what codecs match CN and telephony-event? Where can I find these mappings? m=audio 47904 RTP/AVP 2 101 13 Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20] Thanks Brian! On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian@freeswitch.org> wrote:
Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.
/b
On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
_______________________________________________ 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
13 = CN 101 = Telephony Event 2 = G723-32 /b On Jan 5, 2010, at 11:03 PM, Mark Campbell-Smith wrote:
ahh... thats what I thought originally.
So what codecs match CN and telephony-event? Where can I find these mappings?
m=audio 47904 RTP/AVP 2 101 13
Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
Thanks Brian!
those are not audio codecs CN is just comfort noise and telephone-event is dtmf On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith < mcampbellsmith@gmail.com> wrote:
ahh... thats what I thought originally.
So what codecs match CN and telephony-event? Where can I find these mappings?
m=audio 47904 RTP/AVP 2 101 13
Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
Thanks Brian!
On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian@freeswitch.org> wrote:
Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.
/b
On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Thanks Brian and Anthony. Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ? 2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20] So the only codec they have offered is G723-32 (or G721), which FS only supports as a passthrough codec. Is that correct? On Wed, Jan 6, 2010 at 4:20 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
those are not audio codecs CN is just comfort noise and telephone-event is dtmf
On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
ahh... thats what I thought originally.
So what codecs match CN and telephony-event? Where can I find these mappings?
m=audio 47904 RTP/AVP 2 101 13
Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
Thanks Brian!
On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian@freeswitch.org> wrote:
Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.
/b
On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ 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
no we fully support it, just add it to your config google for iana codec sdp for the reserved numbers and what they mean. On Tue, Jan 5, 2010 at 11:40 PM, Mark Campbell-Smith < mcampbellsmith@gmail.com> wrote:
Thanks Brian and Anthony.
Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20]
So the only codec they have offered is G723-32 (or G721), which FS only supports as a passthrough codec. Is that correct?
On Wed, Jan 6, 2010 at 4:20 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
those are not audio codecs CN is just comfort noise and telephone-event is dtmf
On Tue, Jan 5, 2010 at 11:03 PM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
ahh... thats what I thought originally.
So what codecs match CN and telephony-event? Where can I find these mappings?
m=audio 47904 RTP/AVP 2 101 13
Audio Codec Compare [telephone-event:101:8000:20]/[PCMA:8:8000:20] Audio Codec Compare [CN:13:8000:20]/[G726-32:2:8000:20]
Thanks Brian!
On Wed, Jan 6, 2010 at 3:20 PM, Brian West <brian@freeswitch.org>
wrote:
Lets try this again.. THIS SDP IS NOT from FreeSWITCH its an inbound invite to FreeSWITCH... fix that and you'll be golden.
/b
On Jan 5, 2010, at 9:36 PM, Mark Campbell-Smith wrote:
2010-01-05 18:57:56.845029 [DEBUG] sofia.c:3845 Remote SDP: v=0 o=Sippy 257534956 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47904 RTP/AVP 2 101 13 c=IN IP4 213.50.90.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
They are in fact one in the same please see ITU. /b On Jan 5, 2010, at 11:40 PM, Mark Campbell-Smith wrote:
Thanks Brian and Anthony.
Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20]
So the only codec they have offered is G723-32 (or G721), which FS only supports as a passthrough codec. Is that correct?
The following section in RFC3551 states the same http://tools.ietf.org/html/rfc3551#section-4.5.4 The encodings of G.726 are here denoted as G726-40, G726-32, G726-24, and G726-16. Prior to 1990, G721 described the 32 kbit/s ADPCM encoding, and G723 described the 40, 32, and 16 kbit/s encodings. Thus, G726-32 designates the same algorithm as G721 in RFC 1890. ________________________________ From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Wed, January 6, 2010 9:50:46 AM Subject: Re: [Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20 They are in fact one in the same please see ITU. /b On Jan 5, 2010, at 11:40 PM, Mark Campbell-Smith wrote:
Thanks Brian and Anthony.
Brian: 2 = G723-32 Do you mean G721 instead of G723-32 ?
2010-01-05 18:57:56.845029 [DEBUG] sofia_glue.c:3306 Audio Codec Compare [G721:2:8000:20]/[G726-32:2:8000:20]
So the only codec they have offered is G723-32 (or G721), which FS only supports as a passthrough codec. Is that correct?
_______________________________________________ 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
w00t! :) /b On Jan 6, 2010, at 1:29 AM, Arsen Chaloyan wrote:
The following section in RFC3551 states the same http://tools.ietf.org/html/rfc3551#section-4.5.4
The encodings of G.726 are here denoted as G726-40, G726-32, G726-24, and G726-16. Prior to 1990, G721 described the 32 kbit/s ADPCM encoding, and G723 described the 40, 32, and 16 kbit/s encodings. Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Wed, January 6, 2010 9:50:46 AM Subject: Re: [Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20
They are in fact one in the same please see ITU.
/b
Thanks guys for your response. I'll have a read through the links sent to me. On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs it supports are G726-32, PCMU, PCMA. v=0 o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 31566 RTP/AVP 2 0 8 101 13 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 However, Phonzo responds with the following in the Session Progress message: v=0 o=Sippy 158698636 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 61812 RTP/AVP 2 101 13 c=IN IP4 213.50.91.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Does that make sense? The codec rtpmap only included DTMF ... On Thu, Jan 7, 2010 at 2:04 AM, Brian West <brian@freeswitch.org> wrote:
w00t! :) /b On Jan 6, 2010, at 1:29 AM, Arsen Chaloyan wrote:
The following section in RFC3551 states the same http://tools.ietf.org/html/rfc3551#section-4.5.4
The encodings of G.726 are here denoted as G726-40, G726-32, G726-24, and G726-16. Prior to 1990, G721 described the 32 kbit/s ADPCM encoding, and G723 described the 40, 32, and 16 kbit/s encodings. Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
________________________________ From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Wed, January 6, 2010 9:50:46 AM Subject: Re: [Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20
They are in fact one in the same please see ITU.
/b
_______________________________________________ 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
NO the codec map includes 2 aka g726-32 which you don't have to list in the map if its defined in the standard... what I need is a pcap of the whole process. /b On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:
Thanks guys for your response. I'll have a read through the links sent to me.
On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs it supports are G726-32, PCMU, PCMA.
v=0 o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 31566 RTP/AVP 2 0 8 101 13 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
However, Phonzo responds with the following in the Session Progress message:
v=0 o=Sippy 158698636 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 61812 RTP/AVP 2 101 13 c=IN IP4 213.50.91.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
Does that make sense? The codec rtpmap only included DTMF ...
when they only return 1, that means that is what they want you to use. The call should establish in that case with g726-32 On Wed, Jan 6, 2010 at 6:57 PM, Brian West <brian@freeswitch.org> wrote:
NO the codec map includes 2 aka g726-32 which you don't have to list in the map if its defined in the standard... what I need is a pcap of the whole process.
/b
On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:
Thanks guys for your response. I'll have a read through the links sent to me.
On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs it supports are G726-32, PCMU, PCMA.
v=0 o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 31566 RTP/AVP 2 0 8 101 13 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
However, Phonzo responds with the following in the Session Progress message:
v=0 o=Sippy 158698636 1 IN IP4 80.232.37.178 s=- t=0 0 m=audio 61812 RTP/AVP 2 101 13 c=IN IP4 213.50.91.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
Does that make sense? The codec rtpmap only included DTMF ...
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Mark Campbell-Smith wrote:
ahh... thats what I thought originally.
So what codecs match CN and telephony-event? Where can I find these mappings?
http://www.iana.org/assignments/rtp-parameters is one place to start your journey, but there's enough dynamic assignment, convention, and general etc. to make it an interesting journey. If somebody can set me straight and provide the definitive documentation on payload types, should such actually exist, I'd be most appreciative. -- --Jon Radel jon@radel.com
participants (5)
-
Anthony Minessale
-
Arsen Chaloyan
-
Brian West
-
Jon Radel
-
Mark Campbell-Smith