Hi, My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this? send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565: ------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ Thanks Liam
There is actually a codec list but those are static RTP payload types. You are offering PCMU, PCMA, RFC2833 DTMF, and Comfort Noise See: http://www.iana.org/assignments/rtp-parameters If you need the full SDP for even known static types set verbose_sdp=true in vars.xml. http://wiki.freeswitch.org/wiki/Variable_verbose_sdp Spencer On May 24, 2011, at 7:56 PM, Liam Farr wrote:
Hi,
My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this?
send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565: ------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------
Thanks
Liam _______________________________________________ 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
m=audio 18646 RTP/AVP 0 8 101 13 ^ ^ - these are codecs
Hi,
My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this?
send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:
------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 <http://sip:0800000000@1.2.3.4:5060> SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6 <mailto:sip%3A12345678@9.8.7.6>>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060 <http://sip:0800000000@1.2.3.4:5060>> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060 <http://sip:12345678@9.8.7.6:5060>> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6 <mailto:sip%3A12345678@9.8.7.6>>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
------------------------------------------------------------------------
Thanks
Liam
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-- С уважением, Корнев Е.С. Руководитель группы СПГИ ОАО "СЦК" (343) 378-3100 + 170 Email secured by Check Point
"m=audio 18646 RTP/AVP 0 8 101 13" That is the codec list. 0 = PCMU (G711 u-law) 8 = PCMA (G711 a-law) 101 = DTMF RFC2833 13 = Comfort Noise Numbers 96-127 are dynamic. Anything else is static and defined by IANA ( http://www.iana.org/assignments/rtp-parameters). A dynamic payload type needs a rtpmap line to map the number to the codec name. That is not required for static payload types, because they have a preset rtpmap. FreeSWITCH doesn't add the a=rtpmap lines for static types by default. The advantage of doing so is it makes your packets smaller - less bandwidth usage, and less likely to get a fragmented packet. Some (very few) devices are broken though and require the rtpmap lines for all codecs. These have broken SIP implementations because the RFC states the rtpmap is optional for static types. There is a compatibility option if you're using one of those and really do require it, but changes are you don't: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Steve On 25 May 2011 01:56, Liam Farr <liam@intersys.co.nz> wrote:
Hi,
My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this?
send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565: ------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------
Thanks
Liam
_______________________________________________ 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
Hi, I set <action application="export" data="verbose_sdp=true" /> in my dial plan and this fixed it. My provider is rather strict about going through a testing / certification process for any “new” hardware that they connect to their network, and wanted to see the full codec list in the SIP invite, (even though as you say it’s not required under the RFC’s). Thanks Liam *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Steven Ayre *Sent:* Wednesday, 25 May 2011 6:22 p.m. *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Outbound SIP invites without codec list "m=audio 18646 RTP/AVP 0 8 101 13" That is the codec list. 0 = PCMU (G711 u-law) 8 = PCMA (G711 a-law) 101 = DTMF RFC2833 13 = Comfort Noise Numbers 96-127 are dynamic. Anything else is static and defined by IANA ( http://www.iana.org/assignments/rtp-parameters). A dynamic payload type needs a rtpmap line to map the number to the codec name. That is not required for static payload types, because they have a preset rtpmap. FreeSWITCH doesn't add the a=rtpmap lines for static types by default. The advantage of doing so is it makes your packets smaller - less bandwidth usage, and less likely to get a fragmented packet. Some (very few) devices are broken though and require the rtpmap lines for all codecs. These have broken SIP implementations because the RFC states the rtpmap is optional for static types. There is a compatibility option if you're using one of those and really do require it, but changes are you don't: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Steve On 25 May 2011 01:56, Liam Farr <liam@intersys.co.nz> wrote: Hi, My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this? send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565: ------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ Thanks Liam _______________________________________________ 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
participants (4)
-
"Корнев Е.С."
-
Liam Farr
-
Spencer Thomason
-
Steven Ayre