[asterisk-dev] jingle audio
Landon McDowell
asterisk at ntelos.net
Fri Jul 28 09:41:56 MST 2006
It is worth mentioning that Clod and I were playing with this last night
on my server.
Two way audio works fine with my clients to my server (both with client
NAT and without). He had the same problem with his client to my server
(which again works fine for me). My server wasn't NATted.
Thanks,
Landon McDowell
mogorman wrote:
> Hi Clod
>
> It should be workinking junk-y but you can message me etc, I am gonna be
> playing with jingle all weekend and updating various parts of it if you
> or anyone else wants to help me quash bugs, or give me some bugs to
> quash I should be in #asterisk-dev or just message me at
> mogorman at astjab.org over jabber this weekend.
>
> Mog
> On Thu, 2006-07-27 at 22:36 -0400, Clod Patry wrote:
>
>> im trying to see why audio isn't passing thru chan_jingle when i call
>> from a google talk client.
>>
>> on my dialplan, im just making a Dial(SIP/10), where 10 is my polycom
>> phone.
>>
>> this is what i get:
>> JABBER: asterisk INCOMING: <iq to=" cpatry at gmail.com/asterisk79A5E541"
>> type="set" id="33"
>> from="juliedube01 at gmail.com/Talk.v921D8E248B"><session type="initiate"
>> id="471827166" initiator="juliedube01 at gmail.com/Talk.v921D8E248B"
>> xmlns="http://www.google.com/session"><description
>> xmlns="http://www.google.com/session/phone"><payload-type id="103"
>> name="ISAC"/><payload-type id="97" name="IPCMWB"/><payload-type id="4"
>> name="G723"/><payload-type id="100" name="EG711U"/><payload-type
>> id="101" name="EG711A"/><payload-type id="0"
>> name="PCMU"/><payload-type id="8" name="PCMA"/><payload-type id="13"
>> name="CN"/><payload-type id="102" name="iLBC"/><payload-type id="117"
>> name="red"/><payload-type id="106"
>> name="audio/telephone-event"/></description></session></iq>
>> Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:647 jingle_alloc: The
>> client is guest for alloc
>> caster*CLI>
>> JABBER: asterisk OUTGOING: <iq type='result'
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='33'/>
>> caster*CLI>
>> JABBER: asterisk OUTGOING: <iq
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' type='set'
>> id='aaaae'><session type='candidates' id='471827166' initiator='
>> juliedube01 at gmail.com/Talk.v921D8E248B'
>> xmlns='http://www.google.com/session'><candidate name='rtp'
>> address='127.0.1.1' port='17744' username='034cba614b4c73e8'
>> password='337eb98b143b5bc0' preference='1.00' protocol='udp'
>> type='local' network='0' generation='0'/></session></iq>
>> -- Executing [s at polycom:1] Answer("Jingle/juliedube01-c12b", "")
>> in new stack
>> Jul 27 22:19:32 DEBUG[5499]: chan_jingle.c:398 jingle_answer: Answer!
>> caster*CLI>
>> JABBER: asterisk OUTGOING: <iq type='set'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='aaaaf'><session
>> xmlns='http://www.google.com/session' type='accept'
>> initiator='juliedube01 at gmail.com/Talk.v921D8E248B'
>> id='471827166'><description
>> xmlns='http://www.google.com/session/phone'><payload-type id='0'
>> name='PCMU' xmlns='http://www.google.com/session/phone'/><payload-type
>> id='100' name='EG711U'
>> xmlns='http://www.google.com/session/phone'/><payload-type id='117'
>> name='red' xmlns=' http://www.google.com/session/phone'/><payload-type
>> id='106' name='audio/telephone-event'
>> xmlns='http://www.google.com/session/phone'/><payload-type id='13'
>> name='CN'
>> xmlns='http://www.google.com/session/phone'/></description></session></iq>
>> Jul 27 22:19:32 DEBUG[5284]: channel.c:873 channel_find_locked:
>> Avoiding initial deadlock for channel '0x81fd810'
>> -- Executing [s at polycom:2] Dial("Jingle/juliedube01-c12b",
>> "SIP/10") in new stack
>> Jul 27 22:19:32 DEBUG[5499]: acl.c:213 ast_apply_ha: ##### Testing
>> 192.168.1.10 with 192.168.0.0
>> -- Called 10
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
>> type="set" id="35"
>> from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
>> type="candidates" id="471827166" initiator="
>> juliedube01 at gmail.com/Talk.v921D8E248B"
>> xmlns="http://www.google.com/session"><candidate name="rtp" address="
>> 192.168.1.126" port="2023" username="vk1kh7qZI10Y3ll4"
>> password="1LVHvwyHpSY26BiT" preference="1" protocol="udp" type="local"
>> network="0" generation="0"/></session></iq>
>> Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About
>> to add candidate!
>>
>> JABBER: asterisk OUTGOING: <iq type='result'
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='35'/>
>> Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1333 jingle_parser:
>> Candidate Added!
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
>> type="set" id="36"
>> from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
>> type="candidates" id="471827166"
>> initiator="juliedube01 at gmail.com/Talk.v921D8E248B"
>> xmlns="http://www.google.com/session"><candidate name="rtp"
>> address="70.81.175.205" port="2024" username="5J/dzRIefDQoTt9f"
>> password="3oX373f/FxiftjOd" preference=" 0.9" protocol="udp"
>> type="stun" network="0" generation="0"/></session></iq>
>> Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About
>> to add candidate!
>>
>> JABBER: asterisk OUTGOING: <iq type='result'
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='36'/>
>> Jul 27 22:19:32 DEBUG[5288]: chan_jingle.c:1333 jingle_parser:
>> Candidate Added!
>> -- SIP/10-08203f58 is ringing
>> Jul 27 22:19:32 NOTICE[5499]: chan_jingle.c:1129 jingle_indicate:
>> Don't know how to indicate condition '3'
>> Jul 27 22:19:32 DEBUG[5499]: channel.c:2187 ast_indicate_data: Driver
>> for channel 'Jingle/juliedube01-c12b' does not support indication 3,
>> emulating it
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq type="result" to="
>> cpatry at gmail.com/asterisk79A5E541" id="aaaae"
>> from="juliedube01 at gmail.com/Talk.v921D8E248B"/>
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq type="result"
>> to="cpatry at gmail.com/asterisk79A5E541" id="aaaaf" from="
>> juliedube01 at gmail.com/Talk.v921D8E248B"/>
>> [i pressed answer here, on the polycom]
>> Jul 27 22:19:35 DEBUG[5308]: chan_sip.c:1995 __sip_ack: Acked pending
>> invite 102
>> -- SIP/10-08203f58 answered Jingle/juliedube01-c12b
>> Jul 27 22:19:35 NOTICE[5499]: chan_jingle.c:1129 jingle_indicate:
>> Don't know how to indicate condition '-1'
>> Jul 27 22:19:35 WARNING[5499]: rtp.c:2520 ast_rtp_bridge: Can't find
>> native functions for channel 'Jingle/juliedube01-c12b'
>> -- Native bridging Jingle/juliedube01-c12b and SIP/10-08203f58
>> ended
>> Jul 27 22:19:40 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 192.168.1.3 with 192.168.0.0
>> Jul 27 22:19:40 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 24.37.198.131 with 192.168.0.0
>> Jul 27 22:19:40 DEBUG[5308]: acl.c :213 ast_apply_ha: ##### Testing
>> 192.168.1.3 with 192.168.0.0
>> [i had no audio, just making a show channel to see if its bridged
>> correctly]
>> caster*CLI> show channels
>> Channel Location State Application(Data)
>> SIP/10-08203f58 (None) Up Bridged
>> Call(Jingle/juliedube0
>> Jingle/juliedube01-c s at polycom:2 Up Dial(SIP/10)
>> 2 active channels
>> 1 active call
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
>> type="set" id="37" from="
>> juliedube01 at gmail.com/Talk.v921D8E248B"><session type="candidates"
>> id="471827166" initiator="juliedube01 at gmail.com/Talk.v921D8E248B"
>> xmlns="http://www.google.com/session"><candidate name="rtp"
>> address="192.168.1.126" port="2026" username="EwzANAI7zI+pzjcK"
>> password="hlvEbGin8DP1RBVi" preference="1" protocol="udp" type="local"
>> network="0" generation="1"/></session></iq>
>> Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About
>> to add candidate!
>>
>> JABBER: asterisk OUTGOING: <iq type='result'
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='37'/>
>> Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1333 jingle_parser:
>> Candidate Added!
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
>> type="set" id="38"
>> from="juliedube01 at gmail.com/Talk.v921D8E248B"><session
>> type="candidates" id="471827166"
>> initiator="juliedube01 at gmail.com/Talk.v921D8E248B"
>> xmlns="http://www.google.com/session"><candidate name="rtp"
>> address="70.81.175.205" port="2027" username="RzrYI/z+VcT6V/A0"
>> password="f2bdyUBxrOd58lBT" preference=" 0.9" protocol="udp"
>> type="stun" network="0" generation="1"/></session></iq>
>> Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1331 jingle_parser: About
>> to add candidate!
>>
>> JABBER: asterisk OUTGOING: <iq type='result'
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='38'/>
>> Jul 27 22:19:52 DEBUG[5288]: chan_jingle.c:1333 jingle_parser:
>> Candidate Added!
>> Jul 27 22:19:54 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 192.168.1.3 with 192.168.0.0
>> caster*CLI>
>> caster*CLI>
>> [i pressed hangup here on the polycom]
>> Jul 27 22:20:06 DEBUG[5499]: channel.c:3433 ast_generic_bridge: Didn't
>> get a frame from channel: SIP/10-08203f58
>> Jul 27 22:20:06 DEBUG[5499]: channel.c:3720 ast_channel_bridge: Bridge
>> stops bridging channels Jingle/juliedube01-c12b and SIP/10-08203f58
>> == Spawn extension (polycom, s, 2) exited non-zero on
>> 'Jingle/juliedube01-c12b'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is ''
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is ''
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is 's'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is 'polycom'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is
>> 'Jingle/juliedube01-c12b'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is
>> 'SIP/10-08203f58'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is 'Dial'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is 'SIP/10'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is '2006-07-27
>> 22:19:32'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is '2006-07-27
>> 22:19:32'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is '2006-07-27
>> 22:20:06'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is '34'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is '34'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is 'ANSWERED'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is
>> 'DOCUMENTATION'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is ''
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c:1529
>> pbx_substitute_variables_helper_full: Function result is
>> '1154053172.4'
>> Jul 27 22:20:06 DEBUG[5499]: pbx.c :1529
>> pbx_substitute_variables_helper_full: Function result is ''
>> caster*CLI>
>> JABBER: asterisk OUTGOING: <iq type='set'
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='aaaag'><session
>> type='terminate' id='471827166'
>> initiator='juliedube01 at gmail.com/Talk.v921D8E248B'
>> xmlns='http://www.google.com/session'/></iq>
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq type="result" to="
>> cpatry at gmail.com/asterisk79A5E541" id="aaaag"
>> from="juliedube01 at gmail.com/Talk.v921D8E248B"/>
>> caster*CLI>
>> JABBER: asterisk INCOMING: <iq to="cpatry at gmail.com/asterisk79A5E541"
>> type="set" id="39" from="
>> juliedube01 at gmail.com/Talk.v921D8E248B"><session type="terminate"
>> id="471827166" initiator="juliedube01 at gmail.com/Talk.v921D8E248B"
>> xmlns="http://www.google.com/session"/></iq>
>> Jul 27 22:20:06 DEBUG[5288]: chan_jingle.c:501 jingle_hangup_farend:
>> The client is guest
>> Jul 27 22:20:06 NOTICE[5288]: chan_jingle.c:512 jingle_hangup_farend:
>> Whoa, didn't find call!
>>
>> JABBER: asterisk OUTGOING: <iq type='result'
>> from='cpatry at gmail.com/asterisk79A5E541'
>> to='juliedube01 at gmail.com/Talk.v921D8E248B' id='39'/>
>> Jul 27 22:20:08 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 192.168.1.3 with 192.168.0.0
>> Jul 27 22:20:08 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 24.37.198.131 with 192.168.0.0
>> Jul 27 22:20:08 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 192.168.1.3 with 192.168.0.0
>> Jul 27 22:20:10 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 24.37.198.131 with 192.168.0.0
>> Jul 27 22:20:10 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 192.168.1.3 with 192.168.0.0
>> Jul 27 22:20:10 DEBUG[5308]: acl.c:213 ast_apply_ha: ##### Testing
>> 192.168.1.10 with 192.168.0.0
>> caster*CLI>
>>
>>
>>
>> But even if im just doing a SayDigits(1241), im not hearing it from
>> the google talk client.
>>
>> Any idea on how to solve this issue? Is a problem with RTP and
>> chan_jingle or anything i forgot from my side?
>>
>> --
>> Clod Patry
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