SPA921 Problem Bad Session Description
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs: ------------------------------------------------------------------------ recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397: ------------------------------------------------------------------------ INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it? Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Maxim Balabaev Sent: Monday, October 18, 2010 2:32 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] SPA921 Problem Bad Session Description I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs: ------------------------------------------------------------------------ recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397: ------------------------------------------------------------------------ INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
The SDP is invalid... I suspect your PAP2 has a properly formatted SDP... Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0 /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and "a=rtpmap:0 PCMU/8000" . We WILL NOT make this work because its invalid. Its on you to contact the manufacture of your device and have them fix it. /b On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs: ------------------------------------------------------------------------ recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397: ------------------------------------------------------------------------ INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
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Problem was that fields with some codecs names were blank. Thats why there were "a=rtpmap:0 /8000". Also FS don`t have aliase betwen PCMA and G711.a for ex.
I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it? It`s really old but this is last one on their website.
2010/10/19 Brian West <brian@freeswitch.org>
The SDP is invalid... I suspect your PAP2 has a properly formatted SDP... Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0 /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and "a=rtpmap:0 PCMU/8000" . We WILL NOT make this work because its invalid. Its on you to contact the manufacture of your device and have them fix it.
/b
On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs:
------------------------------------------------------------------------
recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
------------------------------------------------------------------------
INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
------------------------------------------------------------------------
send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
------------------------------------------------------------------------
SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
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set this param in your sofia profile NDLB-allow-bad-iananame=true Then you can use any name you want for the name. but you can't use blank or the low level sip stack will reject it. There is no good in supporting aliases. SIP is fluid enough and the standards that actually work should be preserved. On Tue, Oct 19, 2010 at 9:26 AM, Maxim Balabaev <balabaev.m@gmail.com> wrote:
Problem was that fields with some codecs names were blank. Thats why there were "a=rtpmap:0 /8000". Also FS don`t have aliase betwen PCMA and G711.a for ex.
I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it? It`s really old but this is last one on their website. 2010/10/19 Brian West <brian@freeswitch.org>
The SDP is invalid... I suspect your PAP2 has a properly formatted SDP... Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0 /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and "a=rtpmap:0 PCMU/8000" . We WILL NOT make this work because its invalid. Its on you to contact the manufacture of your device and have them fix it.
/b
On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs:
------------------------------------------------------------------------ recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
------------------------------------------------------------------------ INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
------------------------------------------------------------------------ send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
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Maxim, That SDP looks nasty: a=rtpmap:8 /8000 a=rtpmap:0 /8000 These are static payload types and don't require an rtpmap line (the "offer" is in the m= line). However, when you use an rtpmap you must use the IANA payload type names: PCMU and PCMA in this case. Either of these two SDPs would be valid: v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv -or- v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv Yours isn't :). http://www.iana.org/assignments/rtp-parameters On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m@gmail.com> wrote:
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs: ------------------------------------------------------------------------ recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397: ------------------------------------------------------------------------ INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
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-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
We already told him this when he filed a JIRA on it. It's clearly broken and probably fixed with a firmware update to the device. On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Maxim,
That SDP looks nasty:
a=rtpmap:8 /8000 a=rtpmap:0 /8000
These are static payload types and don't require an rtpmap line (the "offer" is in the m= line). However, when you use an rtpmap you must use the IANA payload type names: PCMU and PCMA in this case. Either of these two SDPs would be valid:
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
-or-
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
Yours isn't :).
http://www.iana.org/assignments/rtp-parameters
On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m@gmail.com> wrote:
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs: ------------------------------------------------------------------------ recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397: ------------------------------------------------------------------------ INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
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-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
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-- 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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Yes. thank you so much. My prev server didn`t require such accurate headers. It`s really cool that fs have such detalied debug. it`s not easy to find that g711x is not acceptible format. 2010/10/19 Anthony Minessale <anthony.minessale@gmail.com>
We already told him this when he filed a JIRA on it. It's clearly broken and probably fixed with a firmware update to the device.
On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Maxim,
That SDP looks nasty:
a=rtpmap:8 /8000 a=rtpmap:0 /8000
These are static payload types and don't require an rtpmap line (the "offer" is in the m= line). However, when you use an rtpmap you must use the IANA payload type names: PCMU and PCMA in this case. Either of these two SDPs would be valid:
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
-or-
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
Yours isn't :).
http://www.iana.org/assignments/rtp-parameters
On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m@gmail.com> wrote:
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs:
recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
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-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
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-- 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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Its not about it not being acceptable. Its about your device putting invalid formatting in the SDP. Has ZERO to do with anything else... your device is busted. /b On Oct 19, 2010, at 9:53 AM, Maxim Balabaev wrote:
Yes. thank you so much. My prev server didn`t require such accurate headers. It`s really cool that fs have such detalied debug. it`s not easy to find that g711x is not acceptible format.
Unfortunately, the model in question (921) has latest fw. For some odd reason, Linksys decided not to issue 5.2.x and 6.x fws for 9x1 models, even though phones are practically same as 9x2 counterparts (PoE is the difference). Bad luck. Maybe you can play with codec name params in web interface of Linksys. On Tue, Oct 19, 2010 at 4:37 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
We already told him this when he filed a JIRA on it. It's clearly broken and probably fixed with a firmware update to the device.
On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Maxim,
That SDP looks nasty:
a=rtpmap:8 /8000 a=rtpmap:0 /8000
These are static payload types and don't require an rtpmap line (the "offer" is in the m= line). However, when you use an rtpmap you must use the IANA payload type names: PCMU and PCMA in this case. Either of these two SDPs would be valid:
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
-or-
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
Yours isn't :).
http://www.iana.org/assignments/rtp-parameters
On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m@gmail.com> wrote:
I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs:
recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
INVITE sip:1001@xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx> Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:022@xxx:5060> Expires: 240 User-Agent: Linksys/SPA921-5.1.8 Content-Length: 386 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp
v=0 o=- 603544 603544 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 /8000 a=rtpmap:0 /8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060 From: <sip:022@xxx>;tag=8f6dbe61c1d0552ao0 To: <sip:1001@xxx>;tag=ytUD271ypvy6r Call-ID: 14d793ed-1abf9d0e@xxx CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0
_______________________________________________ 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
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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I would factory reset it... could just be a bum config. /b On Oct 19, 2010, at 4:27 PM, Ognjen Seslija wrote:
Unfortunately, the model in question (921) has latest fw. For some odd reason, Linksys decided not to issue 5.2.x and 6.x fws for 9x1 models, even though phones are practically same as 9x2 counterparts (PoE is the difference). Bad luck.
Maybe you can play with codec name params in web interface of Linksys.
participants (6)
-
Anthony Minessale
-
Brian West
-
Kristian Kielhofner
-
Maxim Balabaev
-
Nikolay Kondratyev
-
Ognjen Seslija